2017-12-01 19:42 +0000 Asterisk Development Team * asterisk 15.1.3 Released. 2017-11-30 14:38 +0000 [f79ec1ccc8] George Joseph * AST-2017-013: chan_skinny: Call pthread_detach when sess threads end chan_skinny creates a new thread for each new session. In trying to be a good cleanup citizen, the threads are joinable and the unload_module function does a pthread_cancel() and a pthread_join() on any sessions that are active at that time. This has an unintended side effect though. Since you can call pthread_join on a thread that's already terminated, pthreads keeps the thread's storage around until you explicitly call pthread_join (or pthread_detach()). Since only the module_unload function was calling pthread_join, and even then only on the ones active at the tme, the storage for every thread/session ever created sticks around until asterisk exits. * A thread can detach itself so the session_destroy() function now calls pthread_detach() just before it frees the session memory allocation. The module_unload function still takes care of the ones that are still active should the module be unloaded. ASTERISK-27452 Reported by: Juan Sacco Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd 2017-11-10 16:59 +0000 Asterisk Development Team * asterisk 15.1.2 Released. 2017-11-01 11:12 +0000 [f0cebe2321] Ben Ford * res_pjsip: Add to list of valid characters for from_user. Fixes a regression where some characters were unable to be used in the from_user field of an endpoint. Additionally, the backtick was removed from the list of valid characters, since it is not valid, and it was replaced with a single quote, which is a valid character. ASTERISK-27387 Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281 (cherry picked from commit 5b13290522f0dc4312d5dca308bda3fdd636d613) 2017-11-06 16:37 +0000 [7ebf5a9386] Richard Mudgett * res_pjsip_registrar.c: Fix AOR and pjproject group deadlock. One of the patches for ASTERISK_27147 introduced a deadlock regression. When the connection oriented transport shut down, the code attempted to remove the associated contact. However, that same transport had just requested a registration that we hadn't responded to yet. Depending upon timing we could deadlock. * Made send the REGISTER response after we completed processing the request contacts and released the AOR lock to avoid the deadlock. ASTERISK-27391 Change-Id: I89a90f87cb7a02facbafb44c75d8845f93417364 2017-11-08 16:28 +0000 Asterisk Development Team * asterisk 15.1.1 Released. 2017-10-19 13:53 +0000 [fd22ce2ed4] George Joseph * AST-2017-009: pjproject: Add validation of numeric header values Parsing the numeric header fields like cseq, ttl, port, etc. all had the potential to overflow, either causing unintended values to be captured or, if the values were subsequently converted back to strings, a buffer overrun. To address this, new "strto" functions have been created that do range checking and those functions are used wherever possible in the parser. * Created pjlib/include/limits.h and pjlib/include/compat/limits.h to either include the system limits.h or define common numeric limits if there is no system limits.h. * Created strto*_validate functions in sip_parser that take bounds and on failure call the on_str_parse_error function which prints an error message and calls PJ_THROW. * Updated sip_parser to validate the numeric fields. * Fixed an issue in sip_transport that prevented error messages from being properly displayed. * Added "volatile" to some variables referenced in PJ_CATCH blocks as the optimizer was sometimes optimizing them away. * Fixed length calculation in sip_transaction/create_tsx_key_2543 to account for signed ints being 11 characters, not 9. ASTERISK-27319 Reported by: Youngsung Kim at LINE Corporation Change-Id: I48de2e4ccf196990906304e8d7061f4ffdd772ff 2017-10-19 13:35 +0000 [93e6dd60bb] Kevin Harwell * AST-2017-011 - res_pjsip_session: session leak when a call is rejected A previous commit made it so when an invite session transitioned into a disconnected state destruction of the Asterisk pjsip session object was postponed until either a transport error occurred or the event timer expired. However, if a call was rejected (for instance a 488) before the session was fully established the event timer may not have been initiated, or it was canceled without triggering either of the session finalizing states mentioned above. Really the only time destruction of the session should be delayed is when a BYE is being transacted. This is because it's possible in some cases for the session to be disconnected, but the BYE is still transacting. This patch makes it so the session object always gets released (no more memory leak) when the pjsip session is in a disconnected state. Except when the method is a BYE. Then it waits until a transport error occurs or an event timeout. ASTERISK-27345 #close Reported by: Corey Farrell Change-Id: I1e724737b758c20ac76d19d3611e3d2876ae10ed 2017-10-03 16:19 +0000 [6fce3fdedd] Richard Mudgett * AST-2017-010: Fix cdr_object_update_party_b_userfield_cb() buf overrun cdr_object_update_party_b_userfield_cb() could overrun the fixed buffer if the supplied string is too long. The long string could be supplied by external means using the CDR(userfield) function. This may seem reminiscent to AST-2017-001 (ASTERISK_26897) and it is. The earlier patch fixed the buffer overrun for Party A's userfield while this patch fixes the same thing for Party B's userfield. ASTERISK-27337 Change-Id: I0fa767f65ecec7e676ca465306ff9e0edbf3b652 2017-10-30 15:49 +0000 Asterisk Development Team * asterisk 15.1.0 Released. 2017-10-25 20:11 +0000 Asterisk Development Team * asterisk 15.1.0-rc2 Released. 2017-10-22 17:32 +0000 [4a9525a52b] Joshua Colp * res_xmpp: Ensure the connection filter is available. Users of the API that res_xmpp provides expect that a filter be available on the client at all times. When OAuth authentication support was added this requirement was not maintained. This change merely moves the OAuth authentication to after the filter is created, ensuring users of res_xmpp can add things to the filter as needed. ASTERISK-27346 Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886 (cherry picked from commit 0efeb7885a080c92699d6d794c993d2f56e6e55a) 2017-10-25 09:23 +0000 [f534b40813] Ben Ford * http.c: Fix http header send content. Currently ast_http_send barricades a portion of the content that needs to be sent in order to establish a connection for things like the ARI client. The conditional and contents have been changed to ensure that everything that needs to be sent, will be sent. ASTERISK-27372 Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d 2017-10-13 18:19 +0000 Asterisk Development Team * asterisk 15.1.0-rc1 Released. 2017-10-13 12:12 +0000 [f1b646805e] Kevin Harwell * AMI: Increase version number Bump the AMI patch number since the following new addition was made: * Added a new CancelAtxfer action that cancels an attended transfer. Change-Id: I9bac528791bd62ef0e99243903b6bc7a6c7ab182 2017-08-25 08:19 +0000 [eadea6f55b] Thomas Sevestre * features, manager : Add CancelAtxfer AMI action Add action to cancel feature attended transfer with AMI interface ASTERISK-27215 #close Change-Id: Iab8a81362b5a1757e2608f70b014ef863200cb42 2017-10-06 04:55 +0000 [a7d977d986] Daniel Tryba * res_pjsip_session: Prevent user=phone being added to anonimized URIs. Move ast_sip_add_usereqphone to be called after anonymization of URIs, to prevent the user_eq_phone adding "user=phone" to URIs containing a username that is not a phonenumber (RFC3261 19.1.1). An extra call to ast_sip_add_usereqphone on the saved version before anonymization is added to add user=phone" to the PAI. ASTERISK-27047 #close Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6 2017-10-06 05:14 +0000 [3a692949b7] Daniel Tryba * res_pjsip: Prevent "user=phone" being added multiple times to header ast_sip_add_usereqphone adds "user=phone" to the header every time is is called without checking whether the param already exists. Preventing this by searching to string representation of header for "user=phone". ASTERISK-26988 #close Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6 2017-10-10 09:49 +0000 [21e981982f] Tzafrir Cohen * cdr_mysql: avoid releasing a config string Fixes a memory corruption issue after a reload of cdr_mysql. Issue was accidentally included in 747beb1ed159f89a3b58742e4257740b3d6d6bba . ASTERISK-27270 #close Change-Id: I90b6a9d18710c0f9009466370bd5f4bac5d5d12e 2017-10-05 18:12 +0000 [f8da683a16] Richard Mudgett * cdr.c: Defer misc checks. Try to defer some checks until needed in case there is an early exit. Change-Id: Ibc6b34c38a4f60ad4f9b67984b7d070a07257064 2017-10-11 07:03 +0000 [4d518dd748] George Joseph * chan_vpb: Fix a gcc 7 out-of-bounds complaint chan_vpb was trying to use sizeof(*p->play_dtmf), where p->play_dtmf is defined as char[16], to get the length of the array but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf) returns the size of the first array element, which is 1. gcc7 validly complains because the context in which it's used could cause an out-of-bounds condition. Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba 2017-10-09 21:00 +0000 [cd521f166f] Corey Farrell * sorcery: Use ao2_weakproxy to hold list of instances. Store weak proxy objects in instances container. Change-Id: I5a150a4e13cee319d46b5a4654f95a4623a978f8 2017-10-09 21:55 +0000 [d0a49d84ac] Corey Farrell * named_locks: Use ao2_weakproxy_find. Change-Id: I0ce8a1b7101b6caac6a19f83a89f00eaba1e9d9c 2017-10-09 17:51 +0000 [e54238388d] Corey Farrell * astobj2: Add ao2_weakproxy_find function. This function finds a weak proxy in an ao2_container and returns the real object associated with it. Change-Id: I9da822049747275f5961b5c0a7f14e87157d65d8 2017-10-10 15:09 +0000 [0727a8e524] Corey Farrell * astobj2: Run weakproxy callbacks outside of lock. Copy the list of weakproxy callbacks to temporary memory so they can be run without holding the weakproxy lock. Change-Id: Ib167622a8a0f873fd73938f7611b2a5914308047 2017-10-10 12:01 +0000 [9acfcefbd3] Sean Bright * app_originate: Set ORIGINATE_STATUS correctly on failure We were ignoring the return value from ast_pbx_outgoing_exten() and ast_pbx_outgoing_app() which could fail before setting the reason code. This resulted in failures being reported as success. ASTERISK-25266 #close Reported by: Allen Ford Change-Id: Idf16237b7e41b527d2c69c865829128686beeb3b 2017-10-03 15:16 +0000 [7155055b9f] Torrey Searle * contrib/thirdparty/sip_to_pjsip: add additional flag mappings add mappings for udptl redundancy, rtptimeout, and debug flags Change-Id: Ie73cf5c83c05dee01eb9624ede76c1a30225d73a 2017-10-02 16:46 +0000 [a22ac5eee1] Richard Mudgett * cdr.c: Eliminated simple RAII_VAR usages. Change-Id: I150505db307249a962987e7b941bdd369bb91f35 2017-10-09 22:51 +0000 [d106e63258] Corey Farrell * res_pjproject: Fix cleanup of buildopts vector. ASTERISK-27306 Change-Id: I3bed0edf3f55b1d4adcbabb25ec14f11dc766c72 2017-10-03 16:09 +0000 [038759c320] Richard Mudgett * 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 2017-10-02 17:41 +0000 [b48eeff44c] Richard Mudgett * cdr.c: Replace inlined code with ao2_t_replace() Change-Id: I9f424f5282ca7d833592f958d95f1b2bafb549b0 2017-09-29 12:07 +0000 [58c898aa3f] Richard Mudgett * cdr.c: Use current ao2 flag names Change-Id: Ib59d7d2f2a4a822754628f2c48a308d6791a6e6e 2017-09-29 12:31 +0000 [4ad4c5679b] Richard Mudgett * cdr.h: Fix doxygen comments. * Also some misc formatting in cdr.c. Change-Id: Ied89a28802a662c37c43326a1aafdce596e0df4a 2017-09-20 18:36 +0000 [6014a8743d] Richard Mudgett * 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 2017-10-09 08:15 +0000 [737624316d] Sean Bright * res_config_sqlite: Don't enable SQLite CDRs when running 'make samples' Change-Id: I65a5190b2732b2246d67472db70dd37db64ddad4 2017-10-08 14:05 +0000 [335c46185a] David Hajek * 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 2017-10-08 09:11 +0000 [3240ba9336] Alexander Traud * 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 2017-10-07 15:47 +0000 [10d0ffb0e0] Corey Farrell * res_pjsip_session: Fix format_cap leak. ASTERISK-27306 Change-Id: I2c8d3fc148f9f53715c958314e1146f9611741f3 2017-10-05 16:26 +0000 [0c0f6da482] Corey Farrell * 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 2017-10-05 17:59 +0000 [baf331f4bc] Corey Farrell * 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 2017-10-05 19:55 +0000 [70efaf8f3e] Corey Farrell * res_sdp_translator_pjmedia: Fix test unregistration. ASTERISK-27306 Change-Id: Ib3ed47167cb697ab7bd0a56cab589893f491651b 2017-10-05 15:54 +0000 [4ad9ddfc75] Corey Farrell * 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 2017-10-06 02:39 +0000 [bc419d2a9c] Nathan Bruning * app_queue.c: clear moh field in init_queue ASTERISK-27301 #close Change-Id: Ic31361f34e2de3b6470e68fc37205a7711082eba 2017-10-05 20:23 +0000 [622a1de0f0] Corey Farrell * 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 2017-10-02 07:48 +0000 [ba37dd65d3] Daniel Tryba * 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 2017-10-04 10:46 +0000 [9293d85f70] Corey Farrell * 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 2017-09-28 02:56 +0000 [2e1424363a] Benoît Dereck-Tricot * 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 2017-09-28 17:37 +0000 [3344fe4ff9] Richard Mudgett * 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 2017-09-21 14:43 +0000 [ade7253c9d] Richard Mudgett * heap.c: No need to calloc heap pointer array. Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68 2017-09-27 13:45 +0000 [d7e47c88b2] George Joseph * 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 2017-09-28 05:33 +0000 [ce08a5b3f6] Joshua Colp * res_stasis: Add 'video_sfu' as a requested bridge type. This change adds 'video_sfu' as a requested bridge type when creating a bridge. By specifying this a mixing type bridge is created that exchanges video in an SFU fashion. Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606 2017-09-27 11:16 +0000 [7192cacfea] Richard Mudgett * 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 2017-09-26 11:01 +0000 [ab15481553] George Joseph * 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 2017-09-13 21:31 +0000 [1eb84bdb8a] Richard Mudgett * 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 2017-09-26 10:55 +0000 [a3583cabbd] Sean Bright * res_rtp_asterisk: Trim trailing byte off of SDES packet This could have been fixed by subtracting 1 from the final value of 'len' but the way the packet was being constructed was confusing so I took the opportunity to (I think) make it more clear. We were sending 1 extra byte at the end of the SDES RTCP packet which caused Chrome to complain (in its debug log): Too little data (1 byte) remaining in buffer to parse RTCP header (4 bytes). We now send the correct number of bytes. Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b 2017-09-25 13:00 +0000 [a89f14f2d6] Sean Bright * webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file If using a legitimate certificate from a trusted certificate authority, you don't need to provide CA file. Change-Id: I8623973b4209b44889243716d7880274caed8a6d 2017-09-25 13:09 +0000 [c96317c138] Sean Bright * pjproject: Patch to correct STUN FINGERPRINT usage Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6 2017-09-25 12:30 +0000 [3d658e3be4] Kevin Harwell * res_pjsip_session: outgoing call did not offer all configured codecs For some scenarios when an outgoing call was made only a subset of the configured codecs were offered. If the codecs being offered happened to not have a codec supported by the phone then the call would fail. For instance Alice and Bob both are configured in Asterisk for g722 and ulaw( allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains g722 and not both g722 and ulaw, so the call ends. This patch makes it so all the audio codecs configured on the endpoint always get sent, and not just a subset. However priority is given to those codecs that are compatible with the "other side". ASTERISK-27259 #close Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696 2017-09-25 10:59 +0000 [1728e439e8] Richard Mudgett * channel.c: Fix invalid reference in conditionaled out code. ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c 2017-09-25 07:25 +0000 [1fad858934] George Joseph * build: A few gcc 7 error fixes Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec 2017-09-15 02:59 +0000 [d0d4b139fc] Stefan Engström * 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 2017-09-23 12:32 +0000 [f353df0326] Sean Bright * app_stream_echo: Don't echo declined streams Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic Edition after accepting the audio request but declining the video one. Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c 2017-09-22 17:49 +0000 [d80b7e1604] Joshua Colp * res_pjsip_session: Reduce (and improve) SDP renegotiation. When pruning a request to change the topology of a channel be more intelligent about the resulting topology that is actually used for SDP renegotiation. In a case where a stream has not already been negotiated we don't need to renegotiate and offer a declined stream. This can occur if something in Asterisk (such as ConfBridge) requests to add video to a PJSIP channel that has no video codecs configured. In this case since the stream did not already exist we can safely remove the stream from the requested topology, resulting in no renegotiation occurring. In a case where a renegotiation is requested with a codec that is not supported we can reuse the formats of the existing stream if it exists to ensure that the stream continues to flow, instead of removing it. Change-Id: I636540798d55922377318fe619c510fb6ed125fb 2017-09-22 15:29 +0000 [08440df3e3] Kevin Harwell * res_pjsip_session: Don't end session when receiving a 500 on a reinvite During a reinvite, if a remote endpoint error occurs and it returns a 500 the session would end. This patch makes it so the session is not terminated, but continues as it was. The reason for this is because some endpoints may send non session terminating "server errors" like a failed codec negotiation. So in this case instead of ending the call it can hopefully continue. In the case of a real server error the session is already "doomed", will be known soon enough and appropriately ended by Asterisk later. Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5 2017-09-22 10:02 +0000 [9eefce503d] Sean Bright * res_pjsip: Use ast_sip_is_content_type() where appropriate Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d 2017-09-21 09:47 +0000 [962865d4ee] George Joseph * res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint When an INVITE came in with both audio and video streams but there were no audio codecs defined for the endpoint, we weren't declining the audio stream. Since it's usually the first/transport stream, when the video stream was processed and tried to use the transport, it was empty and caused a crash. We now decline the the stream if there are no matching codecs so when the video stream is processed, it's now the first/transport stream and processes normally. Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692 2017-09-16 09:19 +0000 [693c06f8a3] Joshua Colp * bridge: Change participant SFU streams when source streams change. Some endpoints do not like a stream being reused for a new media stream. The frame/jitterbuffer can rely on underlying attributes of the media stream in order to order the packets. When a new stream takes its place without any notice the buffer can get confused and the media ends up getting dropped. This change uses the SSRC change to determine that a new source is reusing an existing stream and then bridge_softmix renegotiates each participant such that they see a new media stream. This causes the frame/jitterbuffer to start fresh and work as expected. ASTERISK-27277 Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07 2017-09-19 05:22 +0000 [365003a444] Rodrigo Ramírez Norambuena * 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 2017-09-20 10:45 +0000 [40de3a12e0] George Joseph * res_pjsip_session: Change some asserts to warning/debug messages There was an issue reported where an SDP received on a 183 Session Progress message caused a crash because the pending streams had already been processed when the OK was received. In that case the pending topology was legitimately NULL. There was an assert for an incorrect number of streams in the topology but not one for topology being NULL. In any case, if you're not in dev-mode the asserts don't do anything and since the scenario is legit, the asserts weren't appropriate anyway. * Changed several asserts to warning or debug messages and return codes as appropriate. ASTERISK-27264 Reported by: Daniel Heckl Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848 2017-09-15 09:43 +0000 [90b68dd54b] Ben Ford * res_pjsip_session: Check for removed stream state. When a sip session is refreshed, the stream topology is looped through, checking each stream for compatible formats. This would cause a crash if the stream state was AST_STREAM_STATE_REMOVED, since the formats would never be set for this stream, causing a NULL value to be returned from ast_stream_get_formats. This commit adds a check for streams with removed states. Also removed a stray semicolon. Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42 2017-09-19 05:44 +0000 [86b897f877] George Joseph * chan_pjsip: Ignore AST_CONTROL_STREAM_TOPOLOGY_CHANGED for now chan_pjsip_indicate was missing a case for the recently added AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an error and causing the call to be hung up instead of just ignoring it. ASTERISK-27260 Reported by: Daniel Heckl Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80 2017-09-07 04:41 +0000 [47b41bd3ae] Jean Aunis * 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 2017-09-18 09:51 +0000 [f5ef14597b] Alexander Traud * res_srtp: lower log level of auth failures 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 2017-09-19 09:34 +0000 [9644f68627] David J. Pryke * 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 2017-09-19 14:28 +0000 [c67ce8ae2d] Richard Mudgett * res_rtp_asterisk.c: Fix bundled SSRC handling. Assertions in the v15+ AST-2017-008 patches found that we were not handling the case if the incoming SDP did not specify the required SSRC attributes for bundled to work. * Be strict on matching SSRC for bundled instances including the parent instance. If the SSRC doesn't match then discard the packet. Bundled has to tell us in the SDP signaling what SSRC to expect. Otherwise, we will not know how to find the bundled instance structure. Change-Id: I152830bbff71c662408909042068fada39e617f9 2017-08-25 17:01 +0000 [a9d6a55ffb] Richard Mudgett * AST-2017-008: Improve RTP and RTCP packet processing. 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 2017-09-19 10:38 +0000 [fbbc4d2433] George Joseph * 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 2017-09-19 07:53 +0000 [2dc1b2f6d9] Joshua Colp * app_confbridge: Only create a channel that records audio. This change makes it so that the conference recorder channel that is created only contains audio formats and an audio stream. This is because the underlying application used by ConfBridge to record, MixMonitor, only allows recording audio. Having additional streams (and in particular a video stream) can result in clients needlessly renegotiating to add a video stream that will never receive video. Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0 2017-09-13 14:14 +0000 [855746fb75] Sean Bright * res_calendar: Plug memory leak and micro-optimization ast_variables_destroy is NULL safe, so there is no need to check its argument before passing it. ASTERISK-25524 #close Reported by: Jesper Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b 2017-09-13 03:46 +0000 [513e0dbb5a] alex * cdr_mysql.c: Apply cdrzone to start and answer Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0 2017-09-13 16:23 +0000 [40722149d7] George Joseph * res_pjsip: Filter out non SIP(S) requests Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460 2017-09-14 07:54 +0000 [7fa11842d8] Joshua Colp * tcptls: Change error message to debug. The Websocket implementation will steal the underlying stream of TCP/TLS sessions. This results in an error message being output about a stream not being present when in reality this is actually fine. This change moves it to a debug message instead. Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094 2017-09-13 14:08 +0000 [31370dc0bd] Sean Bright * res_calendar: Various fixes * The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c 2017-09-13 09:38 +0000 [cca5deb2e5] Sean Bright * chan_rtp: Use μ-law by default instead of signed linear Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3 2017-09-11 05:46 +0000 [752f0533dd] George Joseph * res_pjsip: Add handling for incoming unsolicited MWI NOTIFY A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c 2017-09-08 21:41 +0000 [9b8957ed8a] Richard Mudgett * res_rtp_asterisk.c: Add doxygen to RTCP payload types. Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998 2017-09-11 05:52 +0000 [3552cb511d] George Joseph * alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e 2017-09-10 06:17 +0000 [1d2e58705e] Walter Doekes * res/res_pjsip: Fix localnet checks in pjsip, part 2. In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f 2017-09-05 11:13 +0000 [9620a2bd20] Florian Floimair * alembic: Add support for MS-SQL MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95 2017-09-06 10:50 +0000 [4418fae58b] Sean Bright * app_waitforsilence: Cleanup & don't treat missing frames as 'noise' * WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9 2017-09-06 16:05 +0000 [6dd4ee8926] Scott Griepentrog * chan_sip: when getting sip pvt return failure if not found In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612 2017-09-06 13:38 +0000 [55b0dad30a] Richard Mudgett * stasis/control.c: Fix set_interval_hook() ref leak. Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc 2017-09-01 05:17 +0000 [a824fcc8e8] George Joseph * stasis/control: Fix possible deadlock with swap channel If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3 2017-09-06 05:23 +0000 [07e62b4e0e] Vitezslav Novy * chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e 2017-09-06 07:54 +0000 [2d9aac9f28] George Joseph * alembic: Fix enum creation for dtls_fingerprint Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db 2017-09-05 07:31 +0000 [64f8445c0b] Jacek Konieczny * func_cdr: honour 'u' flag on dummy channel Fixes ${CDR(...,u)} when used in cdr_custom.conf ASTERISK-27165 #close Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83 2017-09-05 11:08 +0000 [e5985c9abe] Florian Floimair * alembic: fix erroneous commit for add_prune_on_boot Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa 2017-09-06 03:02 +0000 [3a418b7ca2] Alexander Traud * res_srtp: Add support for libsrtp2.1. Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with previous versions of libSRTP, this change allows libSRTP 2.1.x as well. ASTERISK-27253 #close Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315 2017-09-05 09:35 +0000 [b7d4a2e32e] Ben Ford * chan_pjsip: Suppress frame warnings. When rtp_keepalive is on for a PJSIP endpoint dialing to another Asterisk instance also using PJSIP, Asterisk will continue to print warning messages about not being able to send frames of a certain type. This suppresses that warning message. Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67 2017-09-05 10:05 +0000 [46bd1290a0] Sean Bright * formats: Restore previous fread() behavior Some formats are able to handle short reads while others are not, so restore the previous behavior for the format modules so that we don't have spurious errors when playing back files. ASTERISK-27232 #close Reported by: Jens T. Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300 2017-09-05 09:16 +0000 [329e111947] Walter Doekes * res/res_pjsip: Standardize/fix localnet checks across pjsip. In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934 2017-09-05 08:39 +0000 [11d013a9df] Joshua Colp * res_pjsip_session: Preserve stream name during renegotiation. Stream names within Asterisk can have meaning so when an externally initiated renegotiation occurs we need to preserve the name of the stream if it already exists. Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596 2017-09-05 07:50 +0000 [3c8b1f40fa] George Joseph * res_calendar*, res_smdi: Move to "extended" support Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d 2017-09-05 05:23 +0000 [9cc4620376] George Joseph * res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel t38_reinvite_response_cb can get called by res_pjsip_session's session_inv_on_tsx_state_changed in situations where session->channel is NULL. If it is, the ast_log warning segfaults because it tries to get the channel name from a NULL channel. * Check session->channel and print "unknown channel" when it's NULL. ASTERISK-27236 Reported by: Ross Beer Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7 2017-09-01 16:17 +0000 [1d33757aa0] Sean Bright * rtp_engine: Prevent possible double free with DTLS config ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89 2017-09-01 13:15 +0000 [1a4c58d1d9] Sean Bright * chan_ooh323: Fix confusing indentation warning ASTERISK-27177 #close Reported by: Tzafrir Cohen Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97 2017-09-01 09:51 +0000 [2872cdfce6] Sean Bright * app_directory: Handle a NULL mailbox without crashing ASTERISK-27241 #close Reported by: David Moore Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6 2017-07-24 10:48 +0000 [5180e2ba27] George Joseph * pjsip_message_ip_updater: Fix issue handling "tel" URIs sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f 2017-07-01 19:24 +0000 [0372157a48] Corey Farrell * AST-2017-006: Fix app_minivm application MinivmNotify command injection An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1 2017-05-22 10:36 +0000 [222d655a2e] Joshua Colp * res_rtp_asterisk: Only learn a new source in learn state. This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c 2017-08-30 07:28 +0000 [6c2c74492d] Joshua Colp * res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. When SDP renegotiation occurs it is possible for an RTP instance to be reused for a new stream, resulting in the remote SSRC changing if it is part of a bundle group. This change allows this and updates its mapping in the current bundle group. ASTERISK-27231 Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490 2017-08-25 21:06 +0000 [033087ba1c] Andre Nazario * chan_pjsip: Add tag info in CHANNEL function Create local_tag and remote_tag in CHANNEL info to get tag from From and To headers of a SIP dialog. ASTERISK-27220 Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524 2017-08-29 14:22 +0000 [2ec26dc635] Richard Mudgett * bridge_native_rtp.c: Fixup native_rtp_framehook() * Fix framehook to test frame type for control frame. * Made framehook exit early if frame type is not a control frame. * Eliminated RAII_VAR in framehook. * Use switch instead of else-if ladder for control frame handling. Change-Id: Ia555fc3600bd85470e3c0141147dbe3ad07c1d18 2017-08-29 09:26 +0000 [9af11c6acd] Sean Bright * confbridge: Handle user hangup during name recording This prevents orphaned CBAnn channels from getting stuck in the bridge. ASTERISK-26994 #close Reported by: James Terhune Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457 2017-08-24 11:45 +0000 [6baf2c26f3] Joshua Colp * core: Reduce video update queueing. A video update frame is used to indicate that a channel with video negotiated should provide a full frame so the decoder decoding the stream is able to do so. In situations where a queue is used to store frames it makes no sense for the queue to contain multiple video update frames. One is sufficient to have a full frame be sent. ASTERISK-27222 Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7 2017-08-25 13:44 +0000 [66388de98c] Sean Bright * voicemail: Fix various abuses of mkstemp mkstemp() returns a unique filename, but appending an extension to that filename does not guarantee uniqueness. Instead, use mkdtemp() and we can put whatever extension we want on the files that we create inside the directory. In the case of app_minivm, we also now properly clean up any temporary files that we create. ASTERISK-20858 #close Reported by: Walter Doekes Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43 2017-08-25 12:20 +0000 [ffb2a781f6] Sean Bright * app_record: Resolve some absolute vs. relative filename bugs If the Record() application is called with a relative filename that includes directories, we were not properly creating the intermediate directories and Record() would fail. Secondarily, updated the documentation for RECORDED_FILE to mention that it does not include a filename extension. Finally, rewrote the '%d' functionality to be a bit more straight forward and less noisy. ASTERISK-16777 #close Reported by: klaus3000 Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2 2017-08-23 10:01 +0000 [5b91992c27] Florian Floimair * alembic: Add dtls_fingerprint column in ps_endpoints table The ps_endpoints table was missing the dtls_fingerprint column introduced with commit adba2a8d7fd. ASTERISK-27168 #close Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd 2017-08-21 04:28 +0000 [c5b89a390c] Torrey Searle * res/res_pjsip_session: allow SDP answer to be regenerated If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1 2017-08-24 09:42 +0000 [6d5e5d0520] Sean Bright * app_queue: Evaluate realtime queues when running dialplan functions ASTERISK-19103 #close Reported by: Jim Van Meggelen Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b 2017-08-23 09:19 +0000 [f18f81cac7] Sean Bright * app_voicemail: Honor escape digits in "greeting only" mode ASTERISK-21241 #close Reported by: Eelco Brolman Patches: Patch uploaded by Eelco Brolman (License 6442) Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe 2017-08-24 08:35 +0000 [e65b4d6525] Sean Bright * res_smdi: Clean up memory leak Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8 2017-08-18 17:37 +0000 [02044e6470] Richard Mudgett * res_pjsip_session.c: Fix crash when declining an active stream. If a previously active stream is declined we could crash because the channel's thread is still using the stream while we are updating the topology in the serializer thread. * Defer removing any declined stream's handler until we have blocked the channel's thread with the channel lock. ASTERISK-27212 Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420 2017-08-16 17:50 +0000 [3e9cd53b5c] Richard Mudgett * bridge_channel.c: Fix FRACK when mapping frames to the bridge. * Add protection checks when mapping streams to the bridge. The channel and bridge may be in the process of updating the stream mapping when a media frame comes in so we may not be able to map the frame at the time. * We need to map the streams to the bridge's stream numbers right before they are written into the bridge. That way we don't have to keep locking/unlocking the bridge and we won't have any synchronization problems before the frames actually go into the bridge. * Protect the deferred queue with the bridge_channel lock. ASTERISK-27212 Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a 2017-08-11 16:31 +0000 [6c555891eb] Richard Mudgett * channel: Fix topology API locking. * ast_channel_request_stream_topology_change() must not be called with any channel locks held. * ast_channel_stream_topology_changed() must be called with only the passed channel lock held. ASTERISK-27212 Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691 2017-08-16 15:22 +0000 [efa6262814] Richard Mudgett * bridge: Fix softmix bridge deadlock. * Fix deadlock in bridge_softmix.c:softmix_bridge_stream_topology_changed() between bridge_channel and channel locks. * The new bridge technology topology change callbacks must be called with the bridge locked. The callback references the bridge channel list, the bridge technology could change, and the bridge stream mapping is updated. ASTERISK-27212 Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be 2017-08-14 12:20 +0000 [0f832afb05] Richard Mudgett * chan_pjsip.c: Fix topology refresh response code accuracy. There are other 1xx and 2xx codes than 100 and 200 respectively. Change-Id: I680db0997343256add1478714f5bf5b5569aee17 2017-08-11 17:06 +0000 [b5987fa191] Richard Mudgett * bridge_softmix.c: Restored softmix_bridge_leave() shortcut exit. Change-Id: I13026cd90954e0265eab94a0faf635a3e11f0e35 2017-08-17 17:07 +0000 [51bcfab4c6] Richard Mudgett * app_confbridge: Document sfu video_mode value. Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204 2017-08-17 17:06 +0000 [a689fcb39f] Richard Mudgett * confbridge.h: Fix doxygen comments. Change-Id: I16133166a85fdb557c66ffcbfe8128d0b4725b0e 2017-08-11 11:40 +0000 [8f707ba8c7] Richard Mudgett * bridge_softmix.c: Remove always true test. Change-Id: I26238df2ff0d0f6dfe95c3aa35da588f1ee71727 2017-08-17 16:46 +0000 [069ed905b4] Sungtae Kim * app_queue: Fix initial hold time queue statistic Fixed to use correct initial value and fixed to use the correct queue info to check the first value. ASTERISK-27204 Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73 2017-08-20 08:15 +0000 [ac6a905bb2] Michael Kuron * res_xmpp: fix inverted return code check in OAuth fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon success and -1 if the function is not available. This commit inverts the return code check so that an error is printed if the module is not loaded and not if it is loaded. ASTERISK-27207 #close Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb 2017-08-17 12:00 +0000 [07c3608c6c] Sean Bright * res_calendar_icalendar: Properly handle recurring events When looking for recurring events, use the correct end time based on the configured 'timeframe.' ASTERISK-27174 #close Reported by: Mark Thompson Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef 2017-08-16 15:43 +0000 [44f23dcae0] George Joseph * Fix downloader not working with curl The codec/dpma downloader wasn't handling curl correctly. The logic that transforms makeopts into a bash-sourceable file wasn't handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was looking for an 'or' command. That logic has been eliminated. Instead of trying to transform and source makeopts, the downloader now calls a make scriptlet to print the value of a specific variable. This way, make handles the ors (or any other make construct that happens to creep into that file). ASTERISK-27202 Reported by: Sean McCord Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99 2017-08-15 13:12 +0000 [f919adac8d] Kevin Harwell * manager: hook event is not being raised When the iostream code went in it introduced a conditional that made it so the hook event was not being raised even if a hook is present. This patch adds a check to see if a hook is present in astman_append. If so then call into the send_string function, which in turn raises the even for specified hook. Also updated the ami hooks unit test, so the test could be automated. ASTERISK-27200 #close Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36 2017-08-15 15:15 +0000 [7d1e1287b6] Richard Mudgett * configure: Check cache for valid pjproject tarball before downloading. On a fresh Asterisk source directory, the bundled pjproject tarball is unconditionally downloaded even if the tarball is already in a specified cache directory. * Made check if the pjproject tarball is valid in the cache directory before downloading the tarball on a fresh source directory. Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5 2017-08-15 11:14 +0000 [d0f6892dff] Richard Mudgett * res_pjsip: Fix prune_on_boot to remove only contacts for the host. * Check that the contact's reg_server matches the host's name before deleting any prune_on_boot contacts. We don't want to delete reliable transport contacts made with other servers if the ps_contacts database table is shared with other servers. Thanks to Ross Beer for pointing out that the original prune logic would delete reliable transport contacts from other servers. ASTERISK-27147 Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0 2017-08-04 09:25 +0000 [4e11c2192f] Andrey Egorov * res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db 2017-08-10 14:18 +0000 [89a92b1f75] Richard Mudgett * STUN/netsock2: Fix some valgrind uninitialized memory findings. * netsock2.c: Test the addr->len member first as it may be the only member initialized in the struct. * stun.c:ast_stun_handle_packet(): The combinded[] local array could get used uninitialized by ast_stun_request(). The uninitialized string gets copied to another location and could overflow the destination memory buffer. These valgrind findings were found for ASTERISK_27150 but are not necessarily a fix for the issue. Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57 2017-08-02 18:44 +0000 [aef38f07d3] Richard Mudgett * res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83 2017-07-31 14:21 +0000 [b9f6697f06] Richard Mudgett * res_pjsip: Remove ephemeral registered contacts on transport shutdown. The fix for the issue is broken up into three parts. This is part two which handles the server side of REGISTER requests when rewrite_contact is enabled. Any registered reliable transport contact becomes invalid when the transport connection becomes disconnected. * Monitor the rewrite_contact's reliable transport REGISTER contact for shutdown. If it is shutdown then the contact must be removed because it is no longer valid. Otherwise, when the client attempts to re-REGISTER it may be blocked because the invalid contact is there. Also if we try to send a call to the endpoint using the invalid contact then the endpoint is not likely to see the request. The endpoint either won't be listening on that port for new connections or a NAT/firewall will block it. * Prune any rewrite_contact's registered reliable transport contacts on boot. The reliable transport no longer exists so the contact is invalid. * Websockets always rewrite the REGISTER contact address and the transport needs to be monitored for shutdown. * Made the websocket transport set a unique name since that is what we use as the ao2 container key. Otherwise, we would not know which transport we find when one of them shuts down. The names are also used for PJPROJECT debug logging. * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state event. Now the global keep_alive_interval option, initially idle shutdown timer, and the server REGISTER contact monitor can work on wetsocket transports. * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. Now initially idle websockets will automatically shutdown. ASTERISK-27147 Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4 2017-07-27 15:36 +0000 [fe630930c9] Richard Mudgett * res_pjsip_transport_management.c: Rename some variables. * Use monitored instead of the misleading keepalive name. Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6 2017-07-28 18:26 +0000 [0de033c9c6] Richard Mudgett * res_pjsip: PJSIP Transport state monitor refactor. The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912 2017-08-09 15:24 +0000 [d0bc274f54] Richard Mudgett * UPGRADE notes: Fixup for the 15 branch Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c 2017-08-10 09:09 +0000 [9d2844fd24] Scott Griepentrog * res_pjsip_messaging: IPv6 receive address needs brackets When handling an incoming SIP MESSAGE, PJSIP attaches the IP address that the message was received from to the message in the variable PJSIP_RECVADDR. When the IP address is IPv6 the :PORT appended results in an unparseable mess. By using an additional bit flag on the pj_sockaddr_print call, the conventional use of brackets around the address is achieved. ASTERISK-27193 #close Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9 2017-07-26 09:17 +0000 [4873e5ff33] Torrey Searle * res_rtp_asterisk: enable rtcp & QOS stats on native bridge Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b 2017-07-28 07:53 +0000 [be8cb75dd2] Torrey Searle * res_rtp_asterisk: Make P2P bridge Asymmetric codec aware Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f 2017-08-08 13:33 +0000 [82f8c19a58] George Joseph * Make --with-pjproject-bundled the default for Asterisk 15 '--with-pjproject-bundled' is now the default when running ./configure. It can be disabled with '--without-pjproject-bundled'. To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. ASTERISK-27189 Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce 2017-08-05 06:36 +0000 [9685a704a6] Joshua Colp * res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we are told to terminate the session we immediately release any media sessions associated with it. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82 2017-08-06 11:15 +0000 [0e352ec510] Joshua Colp * bridge: Fix stream topology/participant locking and video misrouting. This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03 2017-08-05 14:43 +0000 [3dbb1b9f48] Corey Farrell * channel: Fix leak on successful call to chan->tech->requester. joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6 2017-08-04 16:47 +0000 [09995fae53] Kevin Harwell * res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643 2017-08-02 16:08 +0000 [418e062ce5] Corey Farrell * app_privacy: remove unused header asterisk/image.h Change-Id: I56ed530633a642633b18383821069e806c92ae82 2017-08-03 13:13 +0000 [0cbfb95b7f] Tzafrir Cohen * Support GMIME 3.0 Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f 2017-08-03 20:58 +0000 [1a3e08326e] Corey Farrell * Correct some leaks in unit tests. * chan_sip: channel in test_sip_rtpqos_1. * test_config: config hook, config info and global config holder. * test_core_format: format in format_attribute_set_without_interface. * test_stream: unneeded frame duplication. * test_taskprocessor: task_data. Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31 2017-07-26 17:49 +0000 [563264c9b5] Richard Mudgett * res_pjsip_transport_websocket.c: Fix serializer ref leak. Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57 2017-08-02 18:41 +0000 [347e148aa0] Richard Mudgett * res_pjsip_outbound_registration.c: Misc fixes. * Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80 2017-07-31 20:20 +0000 [db6b883087] Richard Mudgett * res_pjsip_nat.c: Remove unnecessary CMP_STOP. Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da 2017-07-31 14:20 +0000 [2cd7bec061] Richard Mudgett * res_pjsip_registrar.c: Remove unnecessary CMP_STOP. Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8 2017-08-02 09:43 +0000 [b0c016cf6e] Kevin Harwell * alembic/res_pjsip: Add "webrtc" configuration option When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b 2017-08-02 14:16 +0000 [cd79a15b2f] Corey Farrell * Fix compile error for old versions of GCC. Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6 2017-08-01 16:36 +0000 [6edefc9480] George Joseph * Revert "res_pjsip_session: Release media resources on session end quicker." This reverts commit 6c374d863c420d2519a52fffba3de5f935a30cdb. Change-Id: Ia89836005df66ef540885afd6f171886de0db442 2017-07-26 11:39 +0000 [8f51443a08] Joshua Colp * res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we know the session is ending we release the media resources immediately. ASTERISK-27110 Change-Id: I3c6a82fe7d2c50b9dc9197cb12ef22f20d337501 2017-07-26 08:48 +0000 [ef39998169] Sean Bright * res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810 2017-07-23 18:34 +0000 [448317ddc7] Joshua Colp * res_pjsip: Add support for dnsmgr to external_media_address. The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2 2017-07-27 20:58 +0000 [7c3cdc81f7] Corey Farrell * Fix compiler warnings on Fedora 26 / GCC 7. GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88 2017-07-26 09:27 +0000 [1078d4f7f6] Sean Bright * app_queue: Add announce-position-only-up option Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6 2017-07-27 06:35 +0000 [63224d5d9c] George Joseph * bundled_pjproject: Improve SSL/TLS error handling OpenSSL has 2 levels or error processing. It's possible for the top layer to return SSL_ERROR_SYSCALL but the lower layer return no error, in which case processing should continue. Only the top layer was being examined though so connections were being torn down when they didn't need to be. This patch adds the examination of the lower level codes, and if they return no errors, allows processing to continue. ASTERISK-27001 Reported-by: Ian Gilmour patches: pjproject-2.6.patch submitted by Ian Gilmour (license 6889) Updated-by: George Joseph and Sauw Ming (Teluu) Merged to upstream pjproject on 7/27/2017 (commit 5631) Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2 2017-06-26 07:52 +0000 [c24cdbf4ae] Torrey Searle * chan_pjsip: add a new function PJSIP_DTMF_MODE This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612 2017-09-01 19:29 +0000 Asterisk Development Team * asterisk 15.0.0-rc1 Released. 2017-07-24 10:48 +0000 [35c8fb1590] George Joseph * pjsip_message_ip_updater: Fix issue handling "tel" URIs sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f 2017-07-01 19:24 +0000 [231ee5e6c6] Corey Farrell * AST-2017-006: Fix app_minivm application MinivmNotify command injection An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1 2017-05-22 10:36 +0000 [ba2c8f1458] Joshua Colp * res_rtp_asterisk: Only learn a new source in learn state. This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c 2017-08-30 07:28 +0000 [663fe3e31f] Joshua Colp * res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. When SDP renegotiation occurs it is possible for an RTP instance to be reused for a new stream, resulting in the remote SSRC changing if it is part of a bundle group. This change allows this and updates its mapping in the current bundle group. ASTERISK-27231 Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490 2017-08-24 11:45 +0000 [dab0389e24] Joshua Colp * core: Reduce video update queueing. A video update frame is used to indicate that a channel with video negotiated should provide a full frame so the decoder decoding the stream is able to do so. In situations where a queue is used to store frames it makes no sense for the queue to contain multiple video update frames. One is sufficient to have a full frame be sent. ASTERISK-27222 Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7 2017-08-14 12:20 +0000 [0a0ef8a1b1] Richard Mudgett * chan_pjsip.c: Fix topology refresh response code accuracy. There are other 1xx and 2xx codes than 100 and 200 respectively. Change-Id: I680db0997343256add1478714f5bf5b5569aee17 2017-08-18 17:37 +0000 [00b10fa1e1] Richard Mudgett * res_pjsip_session.c: Fix crash when declining an active stream. If a previously active stream is declined we could crash because the channel's thread is still using the stream while we are updating the topology in the serializer thread. * Defer removing any declined stream's handler until we have blocked the channel's thread with the channel lock. ASTERISK-27212 Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420 2017-08-16 17:50 +0000 [6acc945533] Richard Mudgett * bridge_channel.c: Fix FRACK when mapping frames to the bridge. * Add protection checks when mapping streams to the bridge. The channel and bridge may be in the process of updating the stream mapping when a media frame comes in so we may not be able to map the frame at the time. * We need to map the streams to the bridge's stream numbers right before they are written into the bridge. That way we don't have to keep locking/unlocking the bridge and we won't have any synchronization problems before the frames actually go into the bridge. * Protect the deferred queue with the bridge_channel lock. ASTERISK-27212 Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a 2017-08-11 16:31 +0000 [efbf0aa8df] Richard Mudgett * channel: Fix topology API locking. * ast_channel_request_stream_topology_change() must not be called with any channel locks held. * ast_channel_stream_topology_changed() must be called with only the passed channel lock held. ASTERISK-27212 Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691 2017-08-16 15:22 +0000 [6bad253669] Richard Mudgett * bridge: Fix softmix bridge deadlock. * Fix deadlock in bridge_softmix.c:softmix_bridge_stream_topology_changed() between bridge_channel and channel locks. * The new bridge technology topology change callbacks must be called with the bridge locked. The callback references the bridge channel list, the bridge technology could change, and the bridge stream mapping is updated. ASTERISK-27212 Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be 2017-08-17 17:07 +0000 [40faa22ce8] Richard Mudgett * app_confbridge: Document sfu video_mode value. Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204 2017-08-16 15:43 +0000 [e52f9b041a] George Joseph * Fix downloader not working with curl The codec/dpma downloader wasn't handling curl correctly. The logic that transforms makeopts into a bash-sourceable file wasn't handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was looking for an 'or' command. That logic has been eliminated. Instead of trying to transform and source makeopts, the downloader now calls a make scriptlet to print the value of a specific variable. This way, make handles the ors (or any other make construct that happens to creep into that file). ASTERISK-27202 Reported by: Sean McCord Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99 2017-08-15 13:12 +0000 [d7b04f22de] Kevin Harwell * manager: hook event is not being raised When the iostream code went in it introduced a conditional that made it so the hook event was not being raised even if a hook is present. This patch adds a check to see if a hook is present in astman_append. If so then call into the send_string function, which in turn raises the even for specified hook. Also updated the ami hooks unit test, so the test could be automated. ASTERISK-27200 #close Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36 2017-08-15 15:15 +0000 [44d316ef4a] Richard Mudgett * configure: Check cache for valid pjproject tarball before downloading. On a fresh Asterisk source directory, the bundled pjproject tarball is unconditionally downloaded even if the tarball is already in a specified cache directory. * Made check if the pjproject tarball is valid in the cache directory before downloading the tarball on a fresh source directory. Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5 2017-08-09 15:24 +0000 [012391920c] Richard Mudgett * UPGRADE notes: Fixup for the 15 branch Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c 2017-08-04 16:47 +0000 [4d3e66eabc] Kevin Harwell * res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643 2017-08-06 11:15 +0000 [71d0424ed5] Joshua Colp * bridge: Fix stream topology/participant locking and video misrouting. This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03 (cherry picked from commit 0e352ec5100331c6a32008acc88d69d0fc58ccdd) 2017-08-08 13:33 +0000 [84600e2682] George Joseph * Make --with-pjproject-bundled the default for Asterisk 15 '--with-pjproject-bundled' is now the default when running ./configure. It can be disabled with '--without-pjproject-bundled'. To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. ASTERISK-27189 Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce 2017-08-05 14:43 +0000 [afd7875e82] Corey Farrell * channel: Fix leak on successful call to chan->tech->requester. joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6 (cherry picked from commit 3dbb1b9f48b0fa23cec2d8e3f94173004da320a4) 2017-08-02 14:16 +0000 [53bba12340] Corey Farrell * Fix compile error for old versions of GCC. Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6 (cherry picked from commit cd79a15b2f9411c6e77f0f6594ff0c46f0ece080) 2017-08-02 09:43 +0000 [c042ad8343] Kevin Harwell * alembic/res_pjsip: Add "webrtc" configuration option When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b (cherry picked from commit b0c016cf6e0bcbe743f4f8286fb9b5ded830ccf7) 2017-08-02 11:44 +0000 Asterisk Development Team * asterisk 15.0.0-beta1 Released. 2017-07-27 20:58 +0000 [aba08692df] Corey Farrell * Fix compiler warnings on Fedora 26 / GCC 7. GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88 2017-07-27 06:35 +0000 [64edb4ed21] George Joseph * bundled_pjproject: Improve SSL/TLS error handling OpenSSL has 2 levels or error processing. It's possible for the top layer to return SSL_ERROR_SYSCALL but the lower layer return no error, in which case processing should continue. Only the top layer was being examined though so connections were being torn down when they didn't need to be. This patch adds the examination of the lower level codes, and if they return no errors, allows processing to continue. ASTERISK-27001 Reported-by: Ian Gilmore Patch-by: Ian Gilmore (pjproject-2.6.patch License 6889) Updated-by: George Joseph and Sauw Ming (Teluu) Merged to upstream pjproject on 7/27/2017 (commit 5631) Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2 2017-07-25 15:17 +0000 [d056f6b2fe] Sean Bright * res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours Change-Id: Ia578ede1a55b21014581793992a429441903278b 2017-07-26 08:48 +0000 [11cd3be506] Sean Bright * res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810 2017-07-26 09:27 +0000 [76270c0f78] Sean Bright * app_queue: Add announce-position-only-up option Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6 2017-06-26 07:52 +0000 [154e74eced] Torrey Searle * chan_pjsip: add a new function PJSIP_DTMF_MODE This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612 2017-07-17 11:01 +0000 [451d86d62e] Joshua Colp * bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45 2017-07-20 08:08 +0000 [2128dc7c87] Sergej Kasumovic * res_stasis_device_state: Unsubscribe should remove old subscriptions Case scenario with Applications ARI: * Once you subscribe to deviceState with Applications REST API, it will be added into subscription pool. * When you unsubscribe it will remove from the device_state_subscription hash table but not from the subscription pool. * When you subscribe again, it will add it to pool again. * Now you will have two subscriptions and you will receive same event twice. This fix should now remove deviceState subscription from pool and it should fix unsubscribe on deviceState. ASTERISK-27130 #close Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4 2017-07-24 13:30 +0000 [927fc6bbd9] Joshua Colp * core: Add VP9 passthrough support. This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc 2017-07-21 15:57 +0000 [9aa4942a49] Matthew Fredrickson * format.h: Fix a few minor errors in comments. A few minor problems were found in comments in format.h. This patch fixes them. Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94 2017-07-23 18:34 +0000 [0219d25e4e] Joshua Colp * res_pjsip: Add support for dnsmgr to external_media_address. The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2 2017-07-19 18:11 +0000 [85c631294a] Richard Mudgett * app_voicemail.c: Allow mailbox entry on authentication retry prompt. The following testsuite voicemail tests were failing to re-enter the mailbox after the first login attempt. tests/apps/voicemail/authenticate_invalid_mailbox tests/apps/voicemail/authenticate_invalid_password The tests were noting the start of the vm-incorrect-mailbox prompt and immediately sending the mailbox for the next login attempt. Since the invalid message playback had to complete before the digits were recognized, the test passed for the wrong reason and added approximately 20 seconds to the test times. * Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox digits like the initial vm-login prompt so the tests are able to enter the intended mailbox. Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8 2017-07-21 14:20 +0000 [e0ad75ec2a] Rusty Newton * Sounds: Update Makefile for Extra sounds 1.5.1 release Incrementing version for the Extra sounds release. 1.5.1 Extra sounds removes two prompts that were moved into the Core packages in the 1.6 Core sounds release. ASTERISK-27142 #close Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7 2017-07-14 13:47 +0000 [715d79b60d] Rusty Newton * say.c: Fix file locations for second, seconds, minute, minutes files The seconds and minutes files have always existed in the base language directory of the Core package. So say.c has always been calling the wrong location (under digits/) for those two files and in the case of second and minute they didn't exist in the Core packages at all. The 1.6 sounds release moves the second and minute files into Core from Extra for the languages that already had them. A future release will include the second and minute files for languages that didn't already have them. This patch just changes all the target locations for second, seconds, minute, and minutes that were under the digits subdir to be under the root of sounds instead. Which is where the sounds will be for some languages after 1.6 sounds and for all languages after a future release. ASTERISK-25810 #close Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 Reported-by: Nicolas Riendeau 2017-06-18 19:24 +0000 [eea9da2f42] Corey Farrell * Core: Add support for systemd socket activation. This change adds support for socket activation of certain SOCK_STREAM listeners in Asterisk: * AMI / AMI over TLS * CLI * HTTP / HTTPS Example systemd units are provided. This support extends to any socket which is initialized using ast_tcptls_server_start, so any unknown modules using this function will support socket activation. Asterisk continues to function as normal if socket activation is not enabled or if systemd development headers are not available during build. ASTERISK-27063 #close Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d 2017-07-21 11:24 +0000 [94de9d3eea] George Joseph * Restore the incorrectly deleted spandspflow2pcap.log Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5 2017-07-21 07:56 +0000 [6239203628] George Joseph * Update make_ari_stubs to correct version Change-Id: I18575b46db48d62edc72f37dc23b4ab22b43a8b1 2017-07-20 09:57 +0000 [6650ae43e1] Sean Bright * corosync: Fix corosync library name in configure.ac Also add new corosync packages to install_prereq. Reported by Travis Ryan in #asterisk-dev Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db 2017-07-20 13:06 +0000 [b172474728] George Joseph * Update MAINLINE_BRANCH to 15 Change-Id: I425d542b600ceabeef2342e9adfeb68c484a043d 2017-07-20 10:52 +0000 [3e8d628c0e] George Joseph * Update AMI and ARI versions for master/15 and update UPDATE.txt AMI goes from 3.2.0 to 4.0.0 ARI goes from 2.0.0 to 3.0.0 Copied UPGRADE.txt -> UPGRADE-15.txt Created new UPGRADE.txt Removed a log file that was accidentally checked in a while ago Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7 2017-07-18 15:04 +0000 [e7d9e42616] Benjamin Keith Ford * pjsip: Increase maximum packet size. The maximum packet size for PJSIP has been increased to handle the multiple streams being added for WebRTC. Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3 2017-07-17 07:19 +0000 [bcd3f65174] Joshua Colp * bridge_softmix: Don't reorder streams on participant leaving. When a participant leaves a bridge while operating in SFU mode their respective stream on every other participant needs to be removed. Leaving the stream out of the new topology results in every stream after it being moved and reordered. This causes problems with clients. Instead simply mark the stream as removed which leaves it in place in the SDP and doesn't reorder or touch any other streams. ASTERISK-27136 Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1 2017-07-16 12:31 +0000 [f48695ce5b] Joshua Colp * bridge_softmix: Use removed stream spots when renegotiating. Streams are never truly removed in SDP, they still occupy a location within the SDP. This location can be reused by another stream if it so chooses. This change takes advantage of this such that if a new stream is needing to be added for a new participant any removed streams are instead replaced first. This reduces the size of the SDP and the number of streams. ASTERISK-27134 Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d 2017-07-16 12:18 +0000 [942ee54b53] Joshua Colp * res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use. This change makes it so that if an RTCP packet is being sent the RTP ICE component is used for sending if RTCP-MUX is in use. ASTERISK-27133 Change-Id: I6200f611ede709602ee9b89501720c29545ed68b 2017-07-14 01:25 +0000 [26f149ab0a] Sergej Kasumovic * app_confbridge: Make sure name recordings are always removed from the filesystem This commit fixes two possible scenarios: * When recording name and if during recording you hangup, file is never removed. This is due to the fact file location is nulled. * When recording name and if you hangup during thank-you prompt, file is never removed. ASTERISK-27123 #close Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625 2017-07-14 01:11 +0000 [d3f5b265c7] Sergej Kasumovic * chan_iax2: On reload make sure to check for existing MWI subscription On every reload of chan_iax2 module, MWI subscription was added, which results in additional taskprocessors being accumulated over time. This commit fixes it by making sure we check for existing subscription first. This was verified with 'core show taskprocessors' CLI command. ASTERISK-27122 #close Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9 2017-07-10 18:17 +0000 [7da6ddda30] Kevin Harwell * res_pjsip: Add "webrtc" configuration option This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd 2017-07-13 15:43 +0000 [3fbb4a0a08] Rusty Newton * Sounds: Update for core sounds 1.6 release Added necessary lines to make the en_NZ language set selectable and to get core sounds 1.6 pulled down. ASTERISK-26807 #close ASTERISK-25816 #close ASTERISK-26274 #close Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4 2017-07-10 14:04 +0000 [78a50b0343] Corey Farrell * core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554 2017-06-30 13:55 +0000 [065c3005ad] Joshua Colp * res_rtp_asterisk / res_pjsip: Add support for BUNDLE. BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11 2017-07-11 09:55 +0000 [8b535a406b] Torrey Searle * res/res_stasis_snoop: generate silence when audiohook returns null Currently when rtp is paused, no packets are written to the recorded audio file, causing the silence to be skipped and recording not properly time aligned. The read handler as been adapted to return a silence frame of the correct size. ASTERISK-27128 #close Change-Id: I2d7f60650457860b9c70907b14426756b058a844 2017-06-22 07:47 +0000 [d42a9cc9dc] Torrey Searle * res/res_pjsip_t38 ensure t38 requests get rejected quickly arm the t38 webhook always, so we can correctly reject a T38 negotiation request when t38 is disabled on a channel Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d 2017-07-12 13:24 +0000 [6b138046e7] Corey Farrell * core: Add digit filtering to ast_waitfordigit_full This adds a parameter to ast_waitfordigit_full which can be used to only stop waiting when certain expected digits are received. Any unexpected DTMF digits are simply ignored. This also creates a new dialplan application WaitDigit. ASTERISK-27129 #close Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9 2017-07-11 04:48 +0000 [b54eb167b4] Holger Hans Peter Freyther * app_playback.c: Use the timezonename parameter In say_date_generic the timezonename parameter is passed but never used. Fix it by passing it to the ast_localtime function. ASTERISK-27124 Change-Id: I63106b8db10426d417d7275f22554a616e92fae4 2017-07-12 15:07 +0000 [e83b9d141a] Sean Bright * basic-pbx: Remove res_pjsip_multihomed from sample config ASTERISK-27127 #close Reported by: HZMI8gkCvPpom0tM Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789 2017-07-11 14:33 +0000 [7f09fd2c2f] Joshua Colp * bridge/core_unreal: Fix SFU bugs with forwarding frames. This change fixes a few things uncovered during SFU testing. 1. Unreal channels incorrectly forwarded video frames when no video stream was present on them. This caused a crash when they were read as the core requires a stream to exist for the underlying media type. The Unreal channel will now ensure a stream exists for the media type before forwarding the frame and if no stream exists then the frame is dropped. 2. Mapping of frames during bridging from the stream number of the underlying channel to the stream number of the bridge was done in the wrong location. This resulted in the frame getting dropped. This mapping now occurs on reading of the frame from the channel. 3. Bridging was using the wrong ast_read function resulting in it living in a non-multistream world. 4. In bridge_softmix when adding new streams to existing channels the wrong stream topology was copied resulting in no streams being added. Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8 2017-07-11 07:26 +0000 [b7a875778a] George Joseph * res_musiconhold: Add kill_escalation_delay, kill_method to class By default, when res_musiconhold reloads or unloads, it sends a HUP signal to custom applications (and all descendants), waits 100ms, then sends a TERM signal, waits 100ms, then finally sends a KILL signal. An application which is interacting with an external device and/or spawns children of its own may not be able to exit cleanly in the default times, expecially if sent a KILL signal, or if it's children are getting signals directly from res_musiconhoild. * To allow extra time, the 'kill_escalation_delay' class option can be used to set the number of milliseconds res_musiconhold waits before escalating kill signals, with the default being the current 100ms. * To control to whom the signals are sent, the "kill_method" class option can be set to "process_group" (the default, existing behavior), which sends signals to the application and its descendants directly, or "process" which sends signals only to the application itself. Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b 2017-07-05 12:44 +0000 [5d86da61a6] Benjamin Keith Ford * manager: Remove AMI "Queues" action. When performing the "Queues" action via AMI, it outputs the same text that the Asterisk CLI outputs when running a "queue show" command, which does not conform with the AMI spec. "QueueStatus" already does what the "Queues" action should do, so instead of correcting the output, the "Queues" action will be removed and "QueueStatus" should be used instead. ASTERISK-27073 #close Reported by: Brian Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8 2017-07-03 07:30 +0000 [d58ef31acd] Tzafrir Cohen * Avoid setting maxfiles for a remote asterisk Setting maxfiles (maximum number of open files) has no practical effect on a remote asterisk (rasterisk, rasterisk -x). It has an ill effect of printing an extra message, which may be annoying in case of -x. ASTERISK-27105 #close Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2 2017-07-05 15:31 +0000 [303f935a50] George Joseph * http.c: Reduce log spam Messages like "fwrite() failed: Connection reset by peer" are no help whatsoever, especially since they can be caused simply by a client disconnecting. * Make those WARNINGs DEBUGs. * Check the return from ast_iostream_printf of headers. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b 2017-07-07 11:19 +0000 [8f72128e66] Benjamin Keith Ford * res_pjsip: Fix crash with from_user containing invalid characters. If the from_user field contains certain characters (like @, {, ^, etc.), PJSIP will return a null value for the URI when attempting to parse it. This causes a crash when trying to dial out through a trunk that contains these invalid characters in its from_user field. This change checks the configuration and ensures that an endpoint will not be created if the from_user contains an invalid character. It also adds a null check to the PJSIP URI parsing as a backup. ASTERISK-27036 #close Reported by: Maxim Vasilev Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0 2017-06-27 19:27 +0000 [03ae8b0105] Richard Mudgett * json.c: Add backtrace log to find 'Invalid UTF-8 string' errors Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929 2017-07-05 13:39 +0000 [9cd8a1df79] Richard Mudgett * res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock. When a message is received on the TURN socket, the code processing the message needs to call into the ICE/STUN session for further processing. This code path locks the TURN group lock then the ICE/STUN group lock. In another thread an ICE/STUN timer can fire off to send a keep alive message over the TURN socket. In this code path, the ICE/STUN group lock is obtained then the TURN group lock is obtained to send the packet. A classic deadlock case if the group locks are not the same. * Made TURN get created using the ICE/STUN session's group lock. NOTE: I was originally concerned that the ICE/STUN session can get recreated by ice_reset_session() for an event like RTCP multiplexing causing a change during SDP negotiation. In this case the TURN group lock would become different. However, TURN is also recreated as part of the ICE/STUN recreation in ice_create() when all known ICE candidates are added to the new ICE session. While the ICE/STUN and TURN sessions are being recreated there is a period where the group locks could be different. ASTERISK-27023 #close Patches: res_rtp_asterisk-turn-deadlock-fix.patch (license #6502) patch uploaded by Michael Walton (modified) Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9 2017-07-06 05:55 +0000 [7a4f577eb7] George Joseph * Fix alembic branches Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187 2017-06-23 11:17 +0000 [1028f64be4] Richard Mudgett * bridge_native_rtp.c: Fix direct media video RTP instance ACL check. The video stream was using the audio stream RTP instance addresses to check if the video RTP gets directed to an allowed direct media Access Control List (ACL) address. There is no guarantee that the video RTP instance uses the same addresses as the audio RTP instance. This looks like it has been a bug since v11 when direct media ACL was first added to chan_sip and then faithfully reproduced through a couple code refactorings into the new bridging architecture. Change-Id: I8ddd56320e0eea769f3ceed3fa5b6bdfb51d681a 2017-07-05 10:29 +0000 [325eeced6a] Sean Bright * core: Remove 'Data Retrieval API' This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83 2017-06-19 11:22 +0000 [d556c67f9f] Rodrigo Ramírez Norambuena * app_queue: Add change priority of call This patch include a feature to change the priority a caller in a queue by CLI and AMI. Change-Id: I55d520d71cc1cefe9a9b81fefaefc14679e96133 2017-07-03 10:59 +0000 [910c05455d] Alexander Traud * chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support). When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was added in any case, because of a local Boolean-negation error of the return value of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was still always added with tlsenable=yes, because the domains were not compared just on the address but also on the port – and TLS is always on a different port than UDP/TCP. ASTERISK-27106 Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c 2017-07-03 10:38 +0000 [4398aa8fa4] Alexander Traud * chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support). Because of a copy-and-paste error when the struct ast_sockaddr changed, tlsbindaddr was not added, when sip.conf contained autodomain=yes; see "show sip domains" on the command-line interface (CLI) of Asterisk. ASTERISK-27106 Change-Id: I3d0957150017c223136968ef1266f275d0d6695e 2017-06-29 13:58 +0000 [950b39a4f5] Sean Bright * app_voicemail: Cleanup ODBC connection handling The primary focus of this patch is adding a missing call to ast_odbc_release_obj(), but is also a general cleanup of the ODBC related code in app_voicemail. ASTERISK-27093 #close Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b 2017-06-30 23:57 +0000 [50ddb56dad] Corey Farrell * channel: Clear channel flag in error branch. Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when ast_read returns NULL. ASTERISK-27100 #close Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d 2017-06-29 18:27 +0000 [b485f6c59c] Richard Mudgett * pjsip_distributor.c: Fix deadlock with TCP type transports. When a SIP message comes in on a transport, pjproject obtains the lock on the transport and pulls the data out of the socket. Unlike UDP, the TCP transport does not allow concurrent access. Without concurrency the transport lock is not released when the transport's message complete callback is called. The processing continues and eventually Asterisk starts processing the SIP message. The first thing Asterisk tries to do is determine the associated dialog of the message to determine the associated serializer. To get the associated serializer safely requires us to get the dialog lock. To send a request or response message for a dialog, pjproject obtains the dialog lock and then obtains the transport lock. Deadlock can result because of the opposite order the locks are obtained. * Fix the deadlock by obtaining the serializer associated with the dialog another way that doesn't involve obtaining the dialog lock. In this case, we use an ao2 container to hold the associated endpoint and serializer. The new locks are held a brief time and won't overlap other existing lock times. ASTERISK-27090 #close Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd 2017-06-29 18:22 +0000 [65a5ac0168] Richard Mudgett * pjsip_distributor.c: Fix unidentified_requests hash functions. The OBJ_SEARCH_xxx defines should not be used as if they were individual bits. They represent a multi-bit enumeration value field. Change-Id: I32abc9a475396dab02402a7014357dd94284e17b 2017-06-29 15:06 +0000 [e7d41050e0] Kevin Harwell * app_stream_echo: misc bug fixes Fixed the following bugs: * calls to stream_echo_write had the last two parameters swapped * ast_read should have been ast_read_stream * added a null check on the frame's subclass format This also resets the update_sent flag upon receiving SRRCHANGE control frame. This will then force a video update. ASTERISK-26997 Change-Id: I6ad7c8253559b800800433c52339e7f5aa583566 2017-06-29 14:56 +0000 [7df7b8a90c] Kevin Harwell * res_rtp_asterisk: trigger source change control frame when dtls is established There needed to be a way to notify handlers upstream that DTLS had been established. This patch makes it so once DTLS has been estalished a source change control frame is put into the read queue. Any handlers can then watch for that frame and trigger off of it. ASTERISK-27096 #close Change-Id: I27ff344f5a8c691a1890dfe3254a4b1a49e7f4a0 2017-06-30 08:31 +0000 [f573e599c0] George Joseph * pjproject_bundled: Allow passing configure options to bundled There wasn't any good way to pass options like --host or --build down to the pjproject configure which makes cross-compiling difficult. * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used to pass arbitrary options to pjproject configure. * Automatically set the pjproject configure --host and --build options to match those supplied for the asterisk configure. ASTERISK-27097 #close Reported-by: Kinsey Moore Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e 2017-06-29 14:50 +0000 [c0c99c7618] George Joseph * chan_pjsip: Fix ability to send UPDATE on COLP When connected_line_method is "invite", we're supposed to determine if the client can support UPDATE and if it can, send UPDATE instead of INVITE to avoid the SDP renegotiation. Not only was pjproject not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing that invite_tsx wasn't NULL which isn't always the case. * Updated chan_pjsip/update_connected_line_information to drop the requirement that invite_tsx isn't NULL. * Submitted patch to pjproject sip_inv.c that sets the PJSIP_INV_SUPPORT_UPDATE flag correctly. * Updated pjsip.conf.sample to clarify what happens when "invite" is specified. ASTERISK-27095 Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560 2017-06-15 03:12 +0000 [fb7247c57c] Torrey Searle * res_pjsip: Add DTMF INFO Failback mode The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91 2017-06-29 03:47 +0000 [ab7d99e62d] Niklas Larsson * app_queue: Add priority to AMI QueueStatus Add priority to callers in AMI QueueStatus response ASTERISK-27092 #close Change-Id: I8d1f737a72c7c38f4cfe1a4ee3ecc0a4f85bd199 2017-05-30 09:12 +0000 [45df25a579] Mark Michelson * chan_pjsip: Add support for multiple streams of the same type. The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7 2017-06-28 09:03 +0000 [642f8356ab] Joshua Colp * res_rtp_asterisk: Fix issues with ICE renegotiation. When re-inviting to add more streams it is possible for the role of existing ICE sessions to be changed to the incorrect value. This results in subsequent refreshes within the sessions getting a role conflict and the ICE session breaking down. This change only sets the role to be the new value if an ICE renegotiation is actually going to happen, otherwise the existing role is preserved. As well if we encounter a situation where a unidirectional ICE negotiation happens and the other side does not send us candidates we will not store any information for sending traffic, even though we know where they are reachable. This change fixes this by using the source of the ICE traffic itself as the target if no candidates are known and we receive some ICE traffic. ASTERISK-27088 Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9 2017-06-27 10:46 +0000 [a48d3e4d31] Torrey Searle * res/res_pjsip_t38: fix incorrect increment of media_count The T38 sdp callback incorrectly has a side effect of incrementing the media_count. This can lead to core dumps. Change-Id: I7bb2f4987de4046ec52cfc34e5ea0662dae32af8 2017-06-08 22:50 +0000 [80e11bd79b] George Joseph * bridge_native_rtp: Keep rtp instance refs on bridge_channel There have been reports of deadlocks caused by an attempt to send a frame to a channel's rtp instance after the channel has left the native bridge and been destroyed. This patch effectively causes the bridge channel to keep a reference to the glue and both the audio and video rtp instances so what gets started will get stopped. ASTERISK-26978 #close Reported-by: Ross Beer Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a 2017-06-27 04:37 +0000 [7827755570] Ivan Poddubny * app_queue: Fix returning to dialplan when a queue is empty The fix for ASTERISK-25665 introduced a regression. The return value of queue_exec used to be 0 in case of leavewhenempty but it was changed to -1 (returned from wait_our_turn and passed transparently by queue_exec), thus leading to hangup instead of returning back to dialplan. This commit resets the value back to 0 in this case, restoring original behavior. ASTERISK-27065 #close Reported by: Marek Cervenka Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac 2017-06-19 17:21 +0000 [0cef7b9d4e] Alexei Gradinari * app_voicemail: IMAP connection control A new global option "imap_poll_logout" was added to specify whether need to disconnect from the IMAP server after polling of mailboxes. ASTERISK-27068 #close Closing IMAP connection after loading mailbox from voicemail.conf ASTERISK-24052 #close Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a 2017-06-21 17:57 +0000 [975e271b01] Richard Mudgett * res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3 2017-06-16 18:08 +0000 [34db4c3993] Alexei Gradinari * res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact Do not need to unsubscribe/subscribe on creating the ednpoint's contact. The modified function create_mwi_subscriptions_for_endpoint adds the subscription only if it does not exist. The subscriptions aren't added for active contacts which are retrieved on startup from realtime if mwi_disable_initial_unsolicited=yes. Because the mwi_contact_added is not called. So the subscriptions also should be created on updating contact. ASTERISK-26230 #close Change-Id: I47e265af9296ca09aa42a316fdacac104148cee4 2017-06-20 16:05 +0000 [27dae55fb6] Kevin Harwell * core_local: local channel data not being properly unref'ed and unlocked In an earlier version of Asterisk a local channel [un]lock all functions were added in order to keep a crash from occurring when a channel hung up too early during an attended transfer. Unfortunately, when a transfer failure occurs and depending on the timing, the local channels sometime do not get properly unlocked and deref'ed after being locked and ref'ed. This happens because the underlying local channel structure gets NULLed out before unlocking. This patch reworks those [un]lock functions and makes sure the values that get locked and ref'ed later get unlocked and deref'ed. ASTERISK-27074 #close Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09 2017-06-20 16:01 +0000 [45a1f4e2ae] Kevin Harwell * bridge: stuck channel(s) after failed attended transfer If an attended transfer failed it was possible for some of the channels involved to get "stuck" because Asterisk was not hanging up the transfer target. This patch ensures Asterisk hangs up the transfer target when an attended transfer failure occurs. ASTERISK-27075 #close Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9 2017-06-19 11:28 +0000 [a7488f8a70] Rodrigo Ramírez Norambuena * cdr: fix mistake spelling of a word for Unanswered. Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df 2017-06-12 16:17 +0000 [d7b6e06abb] Alexei Gradinari * res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact If the endpoint's last contact is deleted unsolicited MWI has to be unsubscribed. ASTERISK-27051 #close Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0 2017-06-16 09:31 +0000 [854a6de819] George Joseph * res_stasis: Plug reference leak on stolen channels When a stasis channel is stolen by another app, the control structure is unreffed but never unlinked from the app_controls container. This causes the channel reference to leak. Added OBJ_UNLINK to the callback in channel_stolen_cb. Also added some additional channel lifecycle debug messages to channel.c. ASTERISK-27059 #close Repoorted-by: George Joseph Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14 2017-06-16 14:56 +0000 [e33bd96638] Matthew Fredrickson * formats/format_g729: Fix typo in comment There was a typo in a comment. This commit is to fix the typo. ASTERISK-27060 #close Change-Id: Ic2699f8dbeaacd58ccb6ec3203e853e1babe3235 2017-06-08 12:28 +0000 [0ad95bc8a0] Frederic LE FOLL * Core/PBX: Deadlock between dialplan execution and application unregistration. Not easy to reproduce, but we have noticed deadlocks when unloading a module while dialplan is handling a request. The deadlock is between : 1) Dialplan execution: pbx_extension_helper() first taking conlock, then pbx_findapp() [when called] asking for lock on apps list. 2) Application unregistration: ast_unregister_application() first taking lock on apps list, then unreference_cached_app() [when called] asking for conlock. As a protection, I suggest to modify ast_unregister_application(), so that it anticipates the need of conlock, before taking the lock on apps list. The side effect is a longer unavailability of conlock when unregistering an application. ASTERISK-27041 Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2 2017-06-12 09:23 +0000 [7a46309d3d] Alexei Gradinari * res_pjsip: New endpoint option "notify_early_inuse_ringing" This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711 2017-06-15 13:48 +0000 [53b7df82f4] Alexei Gradinari * app_voicemail: IMAP logout on reload/unload Closing IMAP connection on module reload or unload. ASTERISK-24052 #close Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525 2017-03-30 09:33 +0000 [9aeab4aced] Jan Friesse * res_corosync: Change thread stack size In Corosync 2.x libraries were changed to use LibQB IPC. Sadly LibQB IPC doesn't support copy-free access to received buffer, so Corosync libraries were rewritten to use stack as buffer. Mostly the needed stack size is quite small, but for all *_dispatch functions, 1MiB is needed. Asterisk function ast_pthread_create_background set stack size for new thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB). This results in Asterisk crash when running with Corosync 2.x. Patch solves this issue by creating it's own version of ast_pthread_create_background which sets stack size to much higher value (actually it's AST_BACKGROUND_STACKSIZE + 3MiB). Another problem may appear when "corosync show members" netconsole command is executed. It is also executed in thread and also has only 500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which again needs at least 1MiB stack. Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x is found, NodeID is displayed instead of IP address. ASTERISK-25370 #close Reported by: mdu113 Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08 2017-06-13 11:33 +0000 [1ac0096512] George Joseph * res_ari: Add "module loaded" check to ari stubs The recent change to make the use of LOAD_DECLINE more consistent caused res_ari to unload itself before declining if the ari.conf file wasn't found. The ari stubs though still tried to use the configuration resulting in segfaults. This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests to see if res_ari is actually loaded and causes the stubs to also decline if it isn't. The macro was then added to the mustache template's "load_module" function. ASTERISK-27026 #close Reported-by: Ronald Raikes Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d 2017-06-15 12:33 +0000 [11ec2945c7] Richard Mudgett * chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read. The construction of the returned string assumed incorrectly that the supplied buffer would always be initialized as an empty string. If it is not an empty string we could overrun the supplied buffer by the length of the non-empty buffer string plus one. It is also theoreticaly possible for the supplied buffer to be overrun by a string terminator during a read operation even if the supplied buffer is an empty string. * Fix the assumption that the supplied buffer would already be an empty string. The buffer is not guaranteed to contain an empty string by all possible callers. * Fix string terminator buffer overrun potential. Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9 2017-06-08 11:38 +0000 [e563a1920e] Richard Mudgett * SDP: Add get/set option calls for RTP sched context per type. Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4 2017-05-11 18:49 +0000 [716abaf33d] Richard Mudgett * SDP: Search for the ice-lite attribute in the right place. * Pulled finding the rtcp-mux attribute flag out of the ICE candidate for loop. Also ordered the RTCP ICE candidate skip test to fail earlier. Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809 2017-05-11 18:46 +0000 [a95584d079] Richard Mudgett * SDP: Set the remote c= line in RTP instance. Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c 2017-06-09 19:03 +0000 [06265b8c8a] Richard Mudgett * stream: Add ast_stream_topology_del_stream() and unit test. Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9 2017-05-11 14:09 +0000 [0fdb99c268] Richard Mudgett * SDP: Add t= line in sdp_create_from_state() Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141 2017-06-14 13:07 +0000 [4797a8bb81] Richard Mudgett * stream: Ignore declined streams for some topology calls. * Made ast_format_cap_from_stream_topology() not include any formats from declined streams. * Made ast_stream_topology_get_first_stream_by_type() ignore declined streams to return the first active stream of the type. * Updated unit tests to check these changes have the expected effect. Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df 2017-06-15 07:32 +0000 [bd16c3c524] Joshua Colp * channel: Fix reference counting in ast_channel_suppress. The ast_channel_suppress function wrongly decremented the reference count of the underlying structure used to keep track of what should be suppressed on a channel if the function was called multiple times on the same channel. This change cleans up the reference counting a bit so this no longer occurs. ASTERISK-27016 Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136 2017-06-14 12:34 +0000 [b8b0b61a24] Richard Mudgett * app_voicemail.c: Fix compile error when IMAP enabled. Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1 2017-06-12 17:55 +0000 [023eede265] Alexei Gradinari * app_voicemail: IMAP logout on MWI unsubscribe Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd 2017-06-14 11:12 +0000 [65ed2ea311] George Joseph * res_pjsip_pubsub: Fix reference to released endpoint destroy_subscription was attempting to get the id of the subscription tree's endpoint after we'd already called ao2_cleanup on it causing a segfault. Moved the cleanup until after the debug statement and since endpoint could also be NULL at this point, check for that as well. ASTERISK-27057 #close Reported-by: Ryan Smith Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678 2017-06-14 08:29 +0000 [ea3f8c6889] George Joseph * res_pjsip_session: Correct inverted test in session_outgoing_nat_hook There was a typo introduced in commit 776ffd77 which was preventing the transport's external media address from being used. ASTERISK-27024 #close Reported-by: Christopher van de Sande patches: patch.diff submitted by Florian Floimair (license 6892) Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27 2017-06-14 08:54 +0000 [88f18faf2a] George Joseph * res_rtp_asterisk: Fix ssrc change for rtcp srtp It looks like there was a copy/paste error in ast_rtp_change_source where if there was a rtcp srtp instance, instead of updating its ssrc we were updating the srtp instance ssrc twice. ASTERISK-27022 #close Reported-by: Michael Walton Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095 2017-06-08 14:38 +0000 [d6386a8f0c] Joshua Colp * bridge: Add a deferred queue. This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415 2017-06-13 14:17 +0000 [9e53c30610] Kevin Harwell * res_pjsip_refer/session: Calls dropped during transfer When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a 2017-06-13 10:47 +0000 [b2fd7e5069] George Joseph * pjproject_bundled: Use the asterisk github mirror for download We now mirror the pjproject tarball and md5 at https://github.com/asterisk/third-party/tree/master/pjproject To improve download reliability, we now get the tarball from our mirror instead of from pjsip.org. ASTERISK-27052 #close Reported-by: 'alex' Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a 2017-06-12 09:57 +0000 [42f738e052] Alexei Gradinari * res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled If sending unsolicited mwi to all endpoints on startup is disabled (mwi_disable_initial_unsolicited=yes) do not need to create subscriptions. If there are many (thousands) realtime endpoints configured with unsolicited mwi and Vociemail Storage configured as ODBC or IMAP there will be huge number of DB/IMAP requests on startup. ASTERISK-26230 #close Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5 2017-06-11 12:06 +0000 [847087a4ff] Sean Bright * codecs.conf.sample: Fix max_bandwidth speling error Reported by Sylvain Boily via asterisk-dev mailing list. Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d 2017-06-08 17:31 +0000 [8d1f54b92e] Jørgen H * res_pjsip_transport_websocket: Add NULL check in get_write_timeout Added check for NULL return value when calling ast_sorcery_retrieve_by_id in function get_write_timeout ASTERISK-27046 Change-Id: I9357717278da631c3a1cb502c412693929b0cb41 2017-06-08 10:54 +0000 [d27168d36f] Guido Falsi * BuildSystem: Add patches to allow building with recent LibreSSL Add some #if defined checks which allow building against LibreSSL. These patchess come from OpenBSD ports: https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/ ASTERISK-27043 #close Reported by: OpenBSD ports Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572 2017-06-06 14:54 +0000 [fcb1a0d7e8] David M. Lee * CFLAGS for BIND8 support Some systems (like macOS) require BIND_8_COMPAT to be defined so that the nameser libraries are, well, BIND8 compatible. Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0 2017-06-08 10:36 +0000 [7b668297f3] Guido Falsi * BuildSystem: Fix build on FreeBSD due to missing crypt.h FreeBSD does not include a crypt.h include file. Definitions for crypt() and crypt_r() are in unistd.h ASTERISK-27042 #close Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e 2017-06-07 15:19 +0000 [5b80496b42] Joshua Colp * chan_pjsip: Update device state when in early media. The chan_pjsip module uses a calculation approach for determining device state. This means that in situations where we would expect device state to change we need to tell the core to query. A scenario that was missed is when early media was signaled. This change adds the notification for the core to query device state when we are told that early media is being provided. ASTERISK-27039 Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f 2017-06-07 14:32 +0000 [e497a76d24] Sean Bright * eventfd: Disable during cross compilation Reported by Lonnie Abelbeck via private e-mail. Change-Id: Icc80f12b8d8d591e14a8e0ed9f1c02cbd193a89b 2017-06-07 11:21 +0000 [19da99df2f] Alexei Gradinari * CHANGES: correct version for a new option 'refer_blind_progress' Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97 2017-06-06 07:04 +0000 [d3e951edf5] Joshua Colp * pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. PJSIP support in Asterisk differs from chan_sip in that it allows media to be sent as-is without transcoding provided the codecs were negotiated in the SDP. This is allowed according to the RFC. Support for this differs quite a lot though and some endpoints do not handle it well. This change extends the 'asymmetric_rtp_codec' option to also cover this case. When set to no (the default) the code behaves as chan_sip does - the best codec is selected and we will only ever send that, unless we change what we are sending if the remote side changes. When set to yes we will send media as-is without transcoding if the codec has been negotiated in the SDP. ASTERISK-26996 Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51 2017-06-06 10:04 +0000 [b3ca24d216] Sean Bright * res_rtp_multicast: Use consistent timestamps when possible When a frame destined for a MulticastRTP channel does not have timing information (such as when an 'originate' is done), we generate the RTP timestamps ourselves without regard to the number of samples we are about to send. Instead, use the same method as res_rtp_asterisk and 'predict' a timestamp given the number of samples. If the difference between the timestamp that we generate and the one we predict is within a specific threshold, use the predicted timestamp so that we end up with timestamps that are consistent with the number of samples we are actually sending. Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f 2017-05-31 10:41 +0000 [861984eac0] Joshua Colp * res_pjsip: Add support for returning only reachable contacts and use it. This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c 2017-06-05 11:27 +0000 [d8802a6a0f] Kevin Harwell * channel: ast_write frame wrongly freed after call to audiohooks ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in ast_write. It would free the frame given to ast_write if the frame returned by ast_audiohook_write_list was different than the given one. The frame give to ast_write should never be freed within that function. It is the caller's resposibility to free the frame after writing (or when it its done with it). By freeing it within ast_write this of course led to some memory corruption problems. This patch makes it so the frame given to ast_write is no longer freed within the function. The frame returned by ast_audiohook_write_list is now subsequently used in ast_write and is freed later. It is freed either after translate if the frame returned by translate is different, or near the end of ast_write prior to function exit. ASTERISK-26973 #close Change-Id: Ic9085ba5f555eeed12f6e565a638c3649695988b 2017-05-31 11:45 +0000 [001f4ddda4] Sean Bright * pbx_builtin: Properly handle hangup during Background Before this patch, when a user hung up during a Background, we would stuff 0xff into a char and attempt a dialplan lookup of it. This caused problems for some realtime engines which interpreted the value as the beginning of an invalid UTF-8 sequence. ASTERISK-19291 #close Reported by: Andrew Nowrot Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358 2017-05-31 04:25 +0000 [f6eeaaafd5] Joshua Colp * channel / app_meetme: Fix parentheses. ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed 2017-05-30 16:07 +0000 [9dce4a947b] Sean Bright * stasis_recording: Correct ast_asprintf error checking ASTERISK-27021 #close Reported by: Tim Morgan Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75 2017-05-28 15:43 +0000 [5c27fe2187] Sean Bright * format: Reintroduce smoother flags In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16 2017-05-24 10:09 +0000 [39d14834f8] Mark Michelson * Confbridge: Add "sfu" video mode to bridge profile options. A previous commit added plumbing to bridge_softmix to allow for an SFU experience with Asterisk. This commit adds an option to app_confbridge that allows for a confbridge to actually make use of the SFU video mode. SFU mode is implemented in a "set it and forget it" kind of way. That is, when the bridge is created, if SFU mode is enabled, then the video mode gets set to SFU and cannot be changed. Future improvements may allow for a hybrid experience (e.g. forward multiple video streams, specifically those of the most recent talkers), but for this addition, no such capability is present. Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020 2017-05-05 11:56 +0000 [2da869408a] Mark Michelson * Add primitive SFU support to bridge_softmix. This sets up the "plumbing" in bridge_softmix to be able to accommodate Asterisk asking as an SFU (selective forwarding unit) for conferences. The way this works is that whenever a channel enters or leaves a conference, all participants in the bridge get sent a stream topology change request. The topologies consist of the channels' original topology, along with video destination streams corresponding to each participants' source video streams. So for instance, if Alice, Bob, and Carol are in the conference, and each supplies one video stream, then the topologies for each would look like so: Alice: Audio, Source video(Alice), Destination Video(Bob), Destination video (Carol) Bob: Audio, Source video(Bob) Destination Video(Alice), Destination video (Carol) Carol: Audio, Source video(Carol) Destination Video(Alice), Destination video (Bob) This way, video that arrives from a source video stream can then be copied out to the destination video streams on the other participants' channels. Once the bridge gets told that a topology on a channel has changed, the bridge constructs a map in order to get the video frames routed to the proper destination streams. This is done using the bridge channel's stream_map. This change is bare-bones with regards to SFU support. Some key features are missing at this point: * Stream limits. This commit makes no effort to limit the number of streams on a specific channel. This means that if there were 50 video callers in a conference, bridge_softmix will happily send out topology change requests to every channel in the bridge, requesting 50+ streams. * Configuration. The plumbing has been added to bridge_softmix, but there has been nothing added as of yet to app_confbridge to enable SFU video mode. * Testing. Some functions included here have unit tests. However, the functionality as a whole has only been verified by hand-tracing the code. * Selectivenss. For a "selective" forwarding unit, this does not currently have any means of being selective. * Features. Presumably, someone might wish to only receive video from specific sources. There are no external-facing functions at the moment that allow for users to select who they receive video from. * Efficiency. The current scheme treats all video streams as being unidirectional. We could be re-using a source video stream as a desetnation, too. But to simplify things on this first round, I did it this way. Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d 2017-05-30 09:34 +0000 [045d7b8cb7] Sean Bright * format_mp3: Re-work menuselect/build issues Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3 to not show up in menuselect), use .PHONY targets when the necessary source files are not present. ASTERISK-23951 Reported by: Tzafrir Cohen Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8 2017-05-30 09:43 +0000 [80206cdc65] George Joseph * test_json: Fix test names with reserved words Some of the test names were actually reserved words (true, false, int, null, string, bool). When the jenkins test results analyzer does its thing it tries to create a map using the test names as keys and fails because they're reserved words. Added "type_" to those test names. Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b 2017-05-26 11:41 +0000 [9c4f63263c] Joshua Colp * manager: Clear the flag on the other channel. During the channel flag audit an incorrect change was done. The flag should be cleared on the second channel. ASTERISK-26469 Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8 2017-05-26 11:15 +0000 [1f136fe885] Sean Bright * res_srtp: Add support for libsrtp2 ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40 2017-05-25 11:10 +0000 [59348aa182] Sean Bright * format_mp3: Don't try to build format_mp3 if we don't have sources ASTERISK-23951 #close Reported by: Tzafrir Cohen Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30 2017-05-23 11:07 +0000 [44c5a144ce] Martin Tomec * Sqlite3: make busy_timeout configurable. Enables runtime configuration of busy_timeout for sqlite databases. Default timeout remains 1000ms. ASTERISK-27014 #close Change-Id: I8921a3aac3c335843be4cb17d2dd0a5c157a36da 2017-05-24 15:50 +0000 [08edd54c1b] George Joseph * unittests: Add a unit test that causes a SEGV and... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed 2017-05-23 15:42 +0000 [d847fe6585] Sean Bright * res_agi: Allow configuration of audio format of EAGI pipe This change allows the format of the EAGI audio pipe to be changed by setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of the loaded formats. ASTERISK-26124 #close Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd 2017-05-23 13:33 +0000 [e2e6baa8d8] Sean Bright * res_agi: Clarify 'RECORD FILE' documentation Documented the 'beep' option in both the parameters list and the command description. ASTERISK-23839 #close Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea 2017-05-23 13:06 +0000 [3dcb3c88aa] Sean Bright * res_agi: Prevent crash when SET VARIABLE called without arguments Explicitly check that the appropriate number of arguments were passed to SET VARIABLE before attempting to reference them. Also initialize the arguments array to zeroes before populating it. ASTERISK-22432 #close Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97 2017-05-23 12:35 +0000 [e490aa3176] Sean Bright * res_agi: Fix malformed AGI usage response If the generated XML documentation for a command does not end with a \n, the postamble of the usage message does not appear on its own line. ASTERISK-25662 #close Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020 2017-05-23 10:06 +0000 [8ae0227cf3] Sean Bright * res_format_attr_h26x: Trim blanks in fmtp attributes Some devices separate format attributes with a semicolon followed by a space, so trim blanks before trying to match them. ASTERISK-27008 #close Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc 2017-05-15 15:03 +0000 [faab058014] Joshua Colp * app_queue: Fix members showing as being in call when not. A change was done which added an 'in_call' flag to queue members that was set to true while talking to an agent. Unfortunately in practice this does not accurately reflect whether they are talking to an agent or not. If a Local channel is involved and a transfer is performed then the app_queue application would incorrectly think the agent was still in a call with the caller. This was done to fix a race condition between an agent becoming available by device state and the checking of the last call information for the wrapup time. There was a small window where the last call information would be the previous value instead of the new one. This change goes about fixing the original issue in a different way by considering the call completed if device state is received which would make the agent available and if they are currently in a call. If this occurs the last call information is updated before the agent becomes available ensuring that old information is not present when checking if the member should be called. This also improves the transfer situation by actually updating and enforcing the wrapup time. ASTERISK-26399 ASTERISK-26400 ASTERISK-26715 ASTERISK-26975 Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea 2017-05-23 05:45 +0000 [36e90952ec] Robert Mordec * app_confbridge: Race between removing and playing name recording while leaving When user leaves a conference, its channel calls async_play_sound_file() in order to play the name announcement and then unlinks the sound file. The async_play_sound_file() function adds a task to conference playback queue, which then runs playback_common() function in a different thread. It leads to a race condition when, in some cases, channel thread may unlink the sound file before playback_common() had a chance to open it. This patch creates a file deletion task, that is queued after playback. ASTERISK-27012 #close Change-Id: I412f7922d412004b80917d4e892546c15bd70dd3 2017-05-22 13:51 +0000 [440ff38c08] Kevin Harwell * res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm When using rtcp mux if an rtcp payload came in it would still use the srtp unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp data was being passed to the rtp unprotect method this would result in an error. This patch ensures that the correct unprotect method is chosen by making sure the passed in rtcp flag is appropriately set when rtcp mux is enabled and an rtcp payload is received. ASTERISK-26979 #close Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241 2017-05-19 10:05 +0000 [0f487978a9] Sean Bright * chan_sip: Better ICE handling for RTCP-MUX If we are offered or are offering RTCP-MUX, don't consider RTCP ICE candidates. This confuses certain browsers (current Firefox for example) and causes intial audio setup delays. ASTERISK-26982 #close Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91 2017-05-12 10:38 +0000 [be4beff3e4] Steve Davies * app_queue: Add QUEUE_RAISE_PENALTY feature Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY, including an extra parameter in queuerules.conf. This value causes lower Agent penalty values to "raise up" so that they can join higher penalty agents and be treated equally after a period of time. ASTERISK-26995 #close Change-Id: If1c6421a983667a5ac4c359f6dac25b212b4c459 2017-04-13 17:17 +0000 [7c0466092c] Mark Michelson * AST-2017-003: Handle zero-length body parts correctly. ASTERISK-26939 #close Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd 2017-04-13 11:14 +0000 [949e9147bf] George Joseph * 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 2017-04-13 17:16 +0000 [2bb98d8fac] Mark Michelson * AST-2017-002: Ensure transaction key buffer is large enough. ASTERISK-26938 #close Change-Id: I266490792fd8896a23be7cb92f316b7e69356413 2017-05-18 16:35 +0000 [4141748e85] Sean Bright * res_hep_rtcp: Add support level to module info Change-Id: I5661478f9cf12d431f730e42be79323b62831e92 2017-05-15 13:26 +0000 [a60d1f3974] Kevin Harwell * app_stream_echo: Added a multi-stream echo application If the channel does not have multi-stream support then this application acts just like app_echo. If it does have multi-stream support then each stream is echoed back to itself (one-to-one). If a "num" is specified, then a new topology is made that contains clones (from the channel's topology) of all media types that are not equal to the given "type". If the media type differs then the first stream matching the "type" is cloned into the new topology and then up to "num" - 1 of the same stream are also cloned into it. Any additional streams from the original topology matching the "type" are subsequently ignored (i.e. not added to the new topology). For this same case when a frame is read from a stream that frame is still echoed back like before, but now that frame is also echoed out to the additional streams that matched on the specified "type". ASTERISK-26997 #close Change-Id: I254144486734178e196c7f590a26ffc13543ff2c 2017-05-15 13:25 +0000 [51375686f7] Kevin Harwell * core/conversions: Added string to unsigned integer and long conversions Added functions that convert a string to an unsigned integer or unsigned long. A couple of unit test were also created to test the routines. The reasons for adding these conversion utilities (and hopefully eventually more) are as follows: * Conversion routines are functionally contained with consistent and better error checking * The function names offer a better description of what is happening * It encourages code reuse for easier bug fixing at a single source * It's simpler to use * It's unit testable For instance, currently in a lot of places when converting to an integer or similar the "sscanf" function is used. When using "sscanf" it may not be immediately clear what's happening as it lacks semantic naming. Limited error checking is usually done as well. For example, most of the time a check is done to make sure the value converted, but does not check for overflows or negative valued conversions when converting unsigned numbers. Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the built in error handling that "strtoul" has. For instance "strtoul" contains overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly more complex in its use. And maybe a bit controversial, but it may be ("big if") potentially slower than "strtoul" in some cases. Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb 2017-05-13 11:40 +0000 [5a7af00e80] Joshua Colp * 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 2017-05-12 21:04 +0000 [30fbed65f1] Richard Mudgett * 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 2017-05-14 00:37 +0000 [6e7b78414f] Rodrigo Ramírez Norambuena * Fix spelling queues.conf.sample file Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee 2017-05-08 13:40 +0000 [93b7f84c1a] Vitezslav Novy * 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 2017-05-08 15:56 +0000 [808f299808] Alexei Gradinari * 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 2017-05-11 00:25 +0000 [045dbcc2d6] Ivan Poddubny * 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 2017-04-27 19:37 +0000 [b8659be9b0] Richard Mudgett * SDP: Make process possible multiple fmtp attributes per rtpmap. Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349 2017-04-28 11:53 +0000 [c2906dfa05] Richard Mudgett * SDP: Remove sdp_state.remote_capabilities The sdp_state.remote_capabilities was only used inside merge_sdps() and subsequent calls to merge_sdps() by re-INVITE's would leak them. Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce 2017-05-05 14:30 +0000 [16785c0908] Richard Mudgett * SDP: Add interface_address to specify our address to use. When we optionally set the interface_address we are forcing the media to go out a specific interface address. This allows us to optionally have the media go out the interface that SIP signalling came in on or if we are configured to have the media always go out a specific address. Change-Id: I160d9fac322a075bd2557b430632544178196189 2017-05-05 14:49 +0000 [367042bd3e] Richard Mudgett * SDP: Explicitly stop a RTP instance before destoying it. * Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream() handle generating disabled/declined streams. * Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not check the offerer side negotiated SDP because that isn't the purpose of this patch and there is much to be done to handle declined/dummy streams. * Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and /main/sdp/sdp_merge_crisscross unit tests. Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31 2017-04-28 19:48 +0000 [be5809fac8] Richard Mudgett * SDP: Rework merge_capabilities(). * Tried to give better variable names. * Made our SDP answer use the offer's RTP payload types as the SDP RFC says we SHOULD. * Updating the local topology now takes the stream format caps. We are likely preparing to send an offer. Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0 2017-04-28 12:30 +0000 [ae7689f093] Richard Mudgett * SDP: Update ast_get_topology_from_sdp() to keep RTP map. * Add failure exits to ast_get_topology_from_sdp(). Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049 2017-05-09 10:34 +0000 [cbbd119c21] Joshua Colp * 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 2017-05-04 17:28 +0000 [10a4439ac9] Joshua Elson * 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 2017-05-05 11:33 +0000 [1a1c86239d] George Joseph * 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 2017-05-09 05:25 +0000 [3c36c29c81] Joshua Colp * 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 2017-05-08 16:11 +0000 [201346fb7d] George Joseph * 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 2017-05-02 18:05 +0000 [56c5c51076] Richard Mudgett * stream: ast_stream_clone() cannot copy the opaque user data. ast_stream_clone() cannot copy the opaque user data stored on a stream. We don't know how to clone the data so it isn't copied into the clone. Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367 2017-05-04 17:32 +0000 [924628812b] Richard Mudgett * netsock2.c: Made get/set addr port avoid potential uninitialized memory. Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647 2017-05-05 08:48 +0000 [4146facfec] Joshua Colp * 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 2017-05-04 16:04 +0000 [0001834157] George Joseph * 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 2017-04-30 16:40 +0000 [c90d81ef51] Joshua Colp * 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 2017-04-25 11:49 +0000 [7b0e3b92fd] Kevin Harwell * bridge_simple: Added support for streams This patch is the first cut at adding stream support to the bridging framework. Changes were made to the framework that allows mapping of stream topologies to a bridge's supported media types. The first channel to enter a bridge initially defines the media types for a bridge (i.e. a one to one mapping is created between the bridge and the first channel). Subsequently added channels merge their media types into the bridge's adding to it when necessary. This allows channels with different sized topologies to map correctly to each other according to media type. The bridge drops any frame that does not have a matching index into a given write stream. For now though, bridge_simple will align its two channels according to size or first to join. Once both channels join the bridge the one with the most streams will indicate to the other channel to update its streams to be the same as that of the other. If both channels have the same number of streams then the first channel to join is chosen as the stream base. A topology change source was also added to a channel when a stream toplogy change request is made. This allows subsystems to know whether or not they initiated a change request. Thus avoiding potential recursive situations. ASTERISK-26966 #close Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163 2017-05-01 13:04 +0000 [008e25def9] Kevin Harwell * 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 2017-05-02 11:34 +0000 [675e058e77] Sean Bright * 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 2017-04-26 16:22 +0000 [cd272da7a8] Richard Mudgett * SDP: Replace SDP telephone_event option with dtmf option The telephone_event option was used as a flag and a bit mapped value in different places when it is a boolean. It is also inadequate to configure the DTMF operation of the RTP instance created for the stream. Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b 2017-04-29 16:11 +0000 [52e4f02b1a] Richard Mudgett * 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 2017-04-28 10:56 +0000 [8170793be6] George Joseph * 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 2017-04-27 16:46 +0000 [48566b8c66] Richard Mudgett * res_sdp_translator_pjmedia.c: Add TODO notes. Change-Id: If27ca61f79accc882c3376d2e876d2b44aa1347b 2017-04-24 18:13 +0000 [ede90e4aa5] Richard Mudgett * SDP: Make SDP translation to/from internal representation more const. Change-Id: I473a174b869728604b37c60853896b0c458bc504 2017-04-20 19:25 +0000 [5c1851cbc0] Richard Mudgett * stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap. Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08 2017-04-24 16:55 +0000 [d71c6e3bfd] Richard Mudgett * SDP: Make ast_sdp_state_set_remote_sdp() return error. Change-Id: I7707c9d872c476d897ff459008652b35142a35e1 2017-04-14 11:52 +0000 [176123e76c] Richard Mudgett * SDP: Misc cleanups (Mostly memory leaks) Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2 2017-04-27 18:15 +0000 [bad091b317] Richard Mudgett * chan_vpb.cc: Fix compile error. Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f 2017-04-26 16:14 +0000 [d6535c0080] Mark Michelson * SDP API: Add SSRC-level attributes RFC 5576 defines how SSRC-level attributes may be added to SDP media descriptions. In general, this is useful for grouping related SSRCes, indicating SSRC-level format attributes, and resolving collisions in RTP SSRC values. These attributes are used widely by browsers during WebRTC communications, including attributes defined by documents outside of RFC 5576. This commit introduces the addition of SSRC-level attributes into SDPs generated by Asterisk. Since Asterisk does not tend to use multiple SSRCs on a media stream, the initial support is minimal. Asterisk includes an SSRC-level CNAME attribute if configured to do so. This at least gives browsers (and possibly others) the ability to resolve SSRC collisions at offer-answer time. In order to facilitate this, the RTP engine API has been enhanced to be able to retrieve the SSRC and CNAME on a given RTP instance. res_rtp_asterisk currently does not provide meaningful CNAME values in its RTCP SDES items, and therefore it currently will always return an empty string as the CNAME value. A task in the near future will result in res_rtp_asterisk generating more meaningful CNAMEs. Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789 2017-04-27 08:02 +0000 [d6b2a58736] George Joseph * 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 2017-04-24 10:59 +0000 [2b22c3c84b] Joshua Colp * channel: Add ability to request an outgoing channel with stream topology. This change extends the ast_request functionality by adding another function and callback to create an outgoing channel with a requested stream topology. Fallback is provided by either converting the requested stream topology into a format capabilities structure if the channel driver does not support streams or by converting the requested format capabilities into a stream topology if the channel driver does support streams. The Dial application has also been updated to request an outgoing channel with the stream topology of the calling channel. ASTERISK-26959 Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6 2017-04-26 14:20 +0000 [c6b757fa05] Kevin Harwell * 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 2017-04-25 11:43 +0000 [cf3429b934] Kevin Harwell * 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 2017-04-26 05:38 +0000 [985a5fd7aa] Joshua Colp * 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 2017-04-26 09:22 +0000 [99dea9ba84] Yasin CANER * res_pjsip_session : fixed wrong From Header number On Re-invite ASTERISK-26964 #close Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302 2017-04-26 08:45 +0000 [858ed60446] George Joseph * 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 2017-04-26 07:58 +0000 [585f9405b1] Thierry Magnien * 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 2017-04-21 12:04 +0000 [f5b67871df] Sean Bright * 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 2017-04-25 07:52 +0000 [199d4776c0] Joshua Colp * 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 2017-04-14 05:21 +0000 [19a79ae12c] Joshua Colp * sdp: Add support for T.38 This change adds a T.38 format which can be used in a stream topology to specify that a UDPTL stream needs to be created. The SDP API has been changed to understand T.38 and create the UDPTL session, add the attributes, and parse the attributes. This change does not change the boundary of the T.38 state machine. It is still up to the channel driver to implement and act on it (such as queueing control frames or reacting to them). ASTERISK-26949 Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7 2017-03-21 15:44 +0000 [32b3e36c68] Mark Michelson * SDP: Ensure SDPs "merge" properly. The gist of this work ensures that when a remote SDP is received, it is merged properly with the local capabilities. The remote SDP is converted into a stream topology. That topology is then merged with the current local topology on the SDP state. That new merged topology is then used to create an SDP. Finally, adjustments are made to RTP instances based on knowledge gained from the remote SDP. There are also a battery of tests in this commit that ensure that some basic SDP merges work as expected. While this may not sound like a big change, it has the property that it caused lots of ancillary changes. * The remote SDP is no longer stored on the SDP state. Biggest reason: there's no need for it. The remote SDP is used at the time it is being set and nowhere else. * Some new SDP APIs were added in order to find attributes and convert generic SDP attributes into rtpmap structures. * Writing tests made me realize that retrieving a value from an SDP options structure, the SDP options needs to be made const. * The SDP state machine was essentially gutted by a previous commit. Initially, I attempted to reinstate it, but I found that as it had been defined, it was not all that useful. What was more useful was knowing the role we play in SDP negotiation, so the SDP state machine has been transformed into an indicator of role. * Rather than storing separate local and joint stream state capabilities, it makes more sense to keep track of current stream state and update it as things change. Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d 2017-04-24 13:16 +0000 [0611f2ca17] Sean Bright * 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 2017-04-17 19:06 +0000 [59203c51cc] Sean Bright * 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 2017-04-21 12:33 +0000 [f1d20c84a1] Richard Mudgett * 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 2017-04-21 12:07 +0000 [835c209445] Richard Mudgett * 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 2017-04-20 02:13 +0000 [b4b1943c5d] Jean Aunis * 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 2017-04-19 15:08 +0000 [c47b3e74d2] Sean Bright * 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 2017-04-19 13:23 +0000 [afad2ffd9f] Richard Mudgett * 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 2017-03-22 16:05 +0000 [d165079cbc] Richard Mudgett * 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 2017-04-19 08:39 +0000 [b8b3380944] Sean Bright * build: Update config.guess and config.sub Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172 2017-04-14 13:52 +0000 [6c0ab9afa7] Sean Bright * 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 2017-04-16 19:59 +0000 [b55d21ad91] George Joseph * 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: I94b97035311eccf52b0101b8590223265a7881d4 2017-04-14 12:51 +0000 [4fb9f5d60e] Sean Bright * 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 2017-04-14 17:32 +0000 [a3e623dd70] Richard Mudgett * Revert "bridging: Ensure successful T.38 negotation" This reverts commit 7819f95791fe0ca0e0cdc417e2687a5900444053. Change-Id: Ib91a7e6c9856f5f41329e42f40ba2394fee861a4 2017-04-14 16:50 +0000 [f6600f2c2e] Sean Bright * 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 2017-04-14 14:36 +0000 [be71be7ed2] Sean Bright * 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 2017-04-12 07:50 +0000 [2e6075c51f] George Joseph * modules: change module LOAD_FAILUREs to LOAD_DECLINES (master) Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258 2017-04-10 05:13 +0000 [72c5f3b0ba] Alexander Traud * res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP. This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compact_headers=yes via the file pjsip.conf. ASTERISK-26932 #close Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689 2017-04-12 07:47 +0000 [6db0939b96] George Joseph * modules: change module LOAD_FAILUREs to LOAD_DECLINES (14) Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e 2017-04-11 11:07 +0000 [747beb1ed1] George Joseph * 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 2017-04-05 06:41 +0000 [7819f95791] Torrey Searle * 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 2017-04-07 08:58 +0000 [7901225261] Torrey Searle * 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 2017-04-07 16:14 +0000 [7312cbe803] Richard Mudgett * 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 2017-04-06 17:31 +0000 [7c37365f03] Richard Mudgett * 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 2017-04-06 18:30 +0000 [8d323c74fa] Richard Mudgett * 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 2017-04-10 11:30 +0000 [5b4e2ec267] Richard Mudgett * res_pjsip: Fix pointer use after unref. Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1 2017-04-06 18:18 +0000 [6f793ac149] Richard Mudgett * 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 2451d4e4550336197ee2e482750cc53f30afa352 ASTERISK-26851 Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc 2017-04-10 17:45 +0000 [d76bc0565c] Richard Mudgett * 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 2017-04-08 03:05 +0000 [2b8dbc9e00] Walter Doekes * samples: Undo removal of include from canonicalize-app-names commit. This include was accidentally removed in changeset Ia79aea64de89531362e993e34230c2044a70aa93. My bad. Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082 2017-04-07 08:35 +0000 [270b485f04] Joshua Colp * 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 2017-04-07 08:06 +0000 [7a46cd7433] Alexander Traud * 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 2017-04-05 09:10 +0000 [e6ae3651b8] Walter Doekes * samples: Canonicalize app names in extensions.conf.sample. This takes care of warnings by ossobv/asterisklint. Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93 2017-04-04 16:20 +0000 [01e9eaf3a6] George Joseph * 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 2017-04-04 11:44 +0000 [fac5115c43] Troy Bowman * app_queue: Log reason for PAUSEALL/UNPAUSEALL We needed the reason for our reporting when agents pause/unpause all of their queues at once. This is a small, simple patch that adds a reason for PAUSEALL and UNPAUSEALL. I have been using it in production for years. ASTERISK-26920 #close Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d 2017-04-05 14:50 +0000 [40e9d5e8b7] George Joseph * sample_config: Add samples for pubsub to pjsip.conf.sample Added: * outbound-publish * resource_list * inbound-publication * asterisk-publication Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a 2017-04-03 15:38 +0000 [f2ee8ac21e] Richard Mudgett * 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 2017-03-27 09:03 +0000 [380973cc47] Corey Farrell * 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 2017-03-25 19:01 +0000 [6c3ae397cb] Daniel Journo * 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 2017-03-31 12:09 +0000 [8e36064109] Corey Farrell * core: Improve/simplify handling of required headers. * Report failures if configure finds a required header is missing. * Deduplicate includes between asterisk.h, astmm.h and compat.h. * Unconditionally include headers in compat.h if required elsewhere. Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5 2017-04-03 13:56 +0000 [a889621b14] Richard Mudgett * 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 2017-04-03 02:30 +0000 [4fc22c7673] Alexander Traud * 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 2017-03-31 16:31 +0000 [48be02c5d8] Joshua Colp * 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 2017-03-31 13:14 +0000 [e8b1bb3041] Richard Mudgett * chan_vpb.cc: Fix compiler error. Added missing channel technology read/write stream callback initialization. Change-Id: I829043a327d987e0d964485dd3d27964bebbd623 2017-03-30 18:28 +0000 [f9695dc057] Corey Farrell * Forward declare 'struct ast_json' in asterisk.h The ast_json structure is used in many Asterisk headers and is often the only part of json.h used. This adds a forward declaration to asterisk.h and removes the include of json.h from many headers. The declaration has been left in endpoints.h and stasis.h to avoid problems with source files that use ast_json functions without directly including json.h. ari.h continues to include json.h as it uses enum ast_json_encoding_format. Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769 2017-03-30 08:11 +0000 [c537f99488] Sean Bright * cdr_pgsql: Fix buffer overflow calling libpq Implement the same buffer size checking done in cel_pgsql. ASTERISK-26896 #close Reported by: twisted Change-Id: Iaacfa1f1de7cb1e9414d121850d2d8c2888f3f48 2017-03-28 13:01 +0000 [a7d94f504f] Walter Doekes * build: Fix deb build issues with fakeroot If DESTDIR is set, don't call ldconfig. Assume that DESTDIR is used to create a binary archive. The ldconfig call should be delegated to the archive postinst script. This fixes the case where fakeroot wraps 'make install' causing $EUID to be 0 even though it doesn't have permission to call ldconfig. The previous logic in configure.ac to detect and correct libdir has been removed as it was not completely accurate. CentOS 64-bit users should again specifiy --libdir=/usr/lib64 when configuring to prevent install to /usr/lib. Updated Makefile:check-old-libdir to check for orphans in lib64 when installing to lib as well as orphans in lib when installing to lib64. Updated Makefile and main/Makefile uninstall targets to remove the orphans using the new logic. ASTERISK-26705 Change-Id: I51739d4a03e60bff38be719b8d2ead0007afdd51 2017-03-27 15:32 +0000 [f3290d6b66] Joshua Colp * sdp: Add support for setting connection address and clean up state. This change cleans up state management for media streams by moving RTP instances into their own session structure and adding additional details that are not relevant to the core (such as connection address). These can live either in the local capabilities or joint capabilities. The ability to set explicit connection address information for the purposes of direct media and NAT has also been added at the global and stream specific level. ASTERISK-26900 Change-Id: If7e5307239a9534420732de11c451a2705b6b681 2017-03-29 10:11 +0000 [5c1ea3ebbd] Sean Bright * astobj2: Prevent potential deadlocks with ao2_global_obj_release The ao2_global_obj_release() function holds an exclusive lock on the global object while it is being dereferenced. Any destructors that run during this time that call ao2_global_obj_ref() will deadlock because a read lock is required. Instead, we make the global object inaccessible inside of the write lock and only dereference it once we have released the lock. This allows the affected destructors to fail gracefully. While this doesn't completely solve the referenced issue (the error message about not being able to create an IQ continues to be shown) it does solve the backtrace spew that accompanied it. ASTERISK-21009 #close Reported by: Marcello Ceschia Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385 2017-03-30 10:18 +0000 [4e5cc70fb4] Corey Farrell * CEL: Remove header declarations of non-existant functions. ast_cel_alloc and ast_cel_destroy do not exist in code, remove them from the headers. Change-Id: I99ce848e2e109e7d61771559f559b9e57973e45c 2017-03-27 11:49 +0000 [f66edcb8b0] Josh Roberson * cel_pgsql.c: Fix buffer overflow calling libpq PQEscapeStringConn() expects the buffer passed in to be an adequitely sized buffer to write out the escaped SQL value string into. It is possible, for large values (such as large values to Dial with a lot of devices) to have more than our 512+1 byte allocation and thus cause libpq to create a buffer overrun. glibc will nicely ABRT asterisk for you, citing a stack smash. Let's only allocate it to be as large as needed: If we have a value, then (strlen(value) * 2) + 1 (as recommended by libpq), and if we have none, just one byte to hold our null will do. ASTERISK-26896 #close Change-Id: If611c734292618ed68dde17816d09dd16667dea2 2017-03-29 08:04 +0000 [e76cc51d5e] Alexander Traud * srtp: Allow zero as tag value for a sRTP Crypto Suite. ASTERISK-25490 #close Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f 2017-03-28 13:10 +0000 [2fe52174de] George Joseph * res_pjsip_config_wizard: Add 2 new parameters to help with proxy config Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff77cd37912b528d9c45b446be091bbc0) (cherry picked from commit 27344675be1941d30508c6e6bd684acdd0791e1a) 2017-03-28 09:29 +0000 [7c0b12dc41] Sean Bright * alembic: Turn off execute bit on non-executable python scripts Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c 2017-03-27 12:37 +0000 [3d8899bacf] Richard Mudgett * Add DTLS sanity check. Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b 2017-03-08 07:24 +0000 [5d938045d4] Joshua Colp * channel: Remove old epoll support and fixed max number of file descriptors. This change removes the old epoll support which has not been used or maintained in quite some time. The fixed number of file descriptors on a channel has also been removed. File descriptors are now contained in a growable vector. This can be used like before by specifying a specific position to store a file descriptor at or using a new API call, ast_channel_fd_add, which adds a file descriptor to the channel and returns its position. Tests have been added which cover the growing behavior of the vector and the new API call. ASTERISK-26885 Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928 2017-03-27 09:35 +0000 [fd204d5c65] Sean Bright * res_musiconhold: Document the 'format' option ASTERISK-26086 #close Reported by: Jens Bürger Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e 2017-03-24 07:43 +0000 [cf6a6226ab] Sean Bright * core: Remove embedded module support This has not worked for some time and is no longer actively maintained. Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99 2017-03-27 08:58 +0000 [d22c678999] Sean Bright * res_musiconhold: Don't chdir() when scanning MoH files There doesn't appear to be any reason that we are chdir'ing in moh_scan_files, and in the event of an Asterisk crash, the core files may not get written because we have changed into a read-only directory. ASTERISK-23996 #close Reported by: Walter Doekes Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354 2017-03-23 09:48 +0000 [d5a8799c4b] Sean Bright * res_xmpp: Use incremental backoff when a read error occurs If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48 2017-03-24 11:29 +0000 [d08c69a9e2] Sean Bright * res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL (44) when a channel is hung up due to an RTP timeout. So do the same when it happens with PJSIP for parity. Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8 2017-03-23 14:01 +0000 [d2f2cdf476] Kevin Harwell * AMI: Updated version Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9 2017-03-23 12:07 +0000 [12dde3b568] Kevin Harwell * pjproject_bundled: raise timeout value used when downloading After configuring Asterisk with '--with-pjproject-bundled' the configure/build process attempts to download pjproject from its download site. Currently, a timeout of 10 seconds is used that will stop the download process if pjproject has not been fully downloaded in that time. For some systems this was not enough time and the process was timing out too early. This patch raises the download timeout value to '60'. Also, this patch fixes another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to DOWNLOAD_TIMEOUT. ASTERISK-26814 #close Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842 2017-03-22 20:33 +0000 [98a88e9ffa] Sean Bright * res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does *not* include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 7 (Not in roster) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 7 (Not in roster) | | invalid@example.org | N/A | Error logged, no return | | invalid@example.org/Valid | N/A | Error logged, no return | +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 1 (Online) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 6 (Offline) | | invalid@example.org | N/A | 7 (Not in roster) | | invalid@example.org/Valid | N/A | 7 (Not in roster) | +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf 2017-03-23 09:45 +0000 [be94105d6d] Sean Bright * res_xmpp: Try to provide useful errors messages from OpenSSL If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4 2017-03-23 05:19 +0000 [ee81ee1f14] Sean Bright * res_xmpp: Fix ref counting issue The only remaining reference to the endpoint is in the endpoints container, and because it is unlinked in ast_endpoint_shutdown, we don't have to explicitly cleanup the endpoint ourselves. Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8 2017-03-23 09:30 +0000 [9493981419] Sean Bright * res_xmpp: Correctly check return value of SSL_connect SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1 2017-03-22 17:32 +0000 [7657c279b5] Sean Bright * res_xmpp: Don't crash when trying to send a message without a connection If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c 2017-03-22 15:40 +0000 [0136ec12a3] Sean Bright * res_xmpp: Include client name in connection related error messages ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9 2017-03-20 13:27 +0000 [9b103e7bea] Kevin Harwell * rtp_engine: allocate RTP dynamic payloads per session Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "yes" in Asterisk 15. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000 Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc 2017-03-21 12:32 +0000 [bb2936f3e4] Sebastian Gutierrez * cdr: Allow setting of user field from 'h' extension The CDR code previously did not allow the user field to be set from the 'h' extension in the dialplan. This change removes that limitation and allows it to be set. ASTERISK-26818 Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6 2017-03-14 16:45 +0000 [6b7697ed48] Richard Begg * res_pjsip_session: Enable RFC3578 overlap dialing support. Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6 2017-03-21 06:59 +0000 [d4fcf196a2] Sean Bright * res_hep: Capture actual transport type in use Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978 2017-03-21 09:57 +0000 [1bf839d44b] Sean Bright * Revert "app_queue: Handle the caller being redirected out of a queue bridge" This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b 2017-03-21 08:26 +0000 [6b4b87787c] Sean Bright * res_pjsip_messaging: Check URI type before dereferencing We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30 2017-03-13 15:21 +0000 [65ad554c98] Joshua Elson * pjsip: prevent memory corruption on creation of xml bodies ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2 2017-03-20 16:27 +0000 [fc794de756] Sean Bright * bridge_softmix: Ignore non-voice frames from translator Some codecs - codec_speex specifically - take voice frames and return other types of frames, like CNG. If we subsequently treat those as voice frames, we'll run into trouble when destroying the frame because of the requirement that each voice frame have an associated format. ASTERISK-26880 #close Reported by: Kirsty Tyerman Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c 2017-03-14 23:49 +0000 [25016a74f8] Aaron An * audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22 2017-03-18 12:30 +0000 [fc71c18a9b] Sean Bright * thread safety: Don't use getprotobyname() POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48 2017-03-19 13:26 +0000 [516e028b44] Sean Bright * res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret We are currently passing in the capacity of the read buffer instead of the number of bytes that we actually read off the wire. Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36 2017-03-14 09:27 +0000 [79069f8ccb] Robert Mordec * app_queue: Member stuck as pending after forwarding previous call from queue Queue member will get stuck in pending_members if queue calls a device that is different from the one observed for state changes. This patch removes members from pending_members as a result of channel stasis events such as blind or attended transfers and hangup. ASTERISK-26862 #close Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727 2017-02-22 23:26 +0000 [8cb4f9cea1] Richard Mudgett * CHANNEL(callid): Give dialplan access to the callid. * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f 2017-03-16 08:42 +0000 [c13ea6080e] Sean Bright * app_queue: Fix locking behavior in stasis message handlers The queue_stasis_data structure contains various mutable fields that require appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and 'caller_uniqueid' fields need to be locked when read from or written to. Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088 2017-03-07 19:28 +0000 [15aa3c0a23] Sean Bright * chan_sip: Add rtcp-mux support ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639 2017-03-16 16:50 +0000 [57656e2b5b] Richard Mudgett * app_confbridge: Fix ConfbridgeTalking AMI event description. Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705 2017-03-16 16:37 +0000 [82982a191c] Richard Mudgett * res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. struct ast_rtcp does not define the dtls member if SRTP is not enabled. ASTERISK-26732 Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e 2017-03-16 15:45 +0000 [49b1f1ca16] Richard Mudgett * res_pjsip_sdp_rtp.c: Fix cut-n-paste error We were inadvertenly referencing the cos_video option to determine if we should set the tos_audio and cos_audio value on the RTP instance. Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0 2017-03-16 10:39 +0000 [e6dc28b78f] Matt Jordan * res/res_pjsip_session: Only check localnet if it is defined If local_net is not defined on a transport, transport_state->localnet will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in this case, causing the external_media_address, if set, to be skipped. This patch causes us to only check if we are sending within a network if local_net is defined. ASTERISK-26879 #close Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb 2017-03-14 16:22 +0000 [44568fc712] Richard Begg * res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport Currently a wildcard address is used for the local RTP socket, which will not always result in the same address as used by the SIP socket (e.g. if explicit transport addresses are configured). Use the transport's host address when binding new local RTP sockets if available. ASTERISK-26851 Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a 2017-03-07 08:33 +0000 [5013d8f5d3] George Joseph * res_pjsip: Symmetric transports A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f 2017-03-16 09:07 +0000 [68749a9fa7] Joshua Colp * res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped. This change removes an assumption that when DTLS is stopped an RTCP session will be present on the RTP session. This is not always the case. ASTERISK-26732 Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611 2017-03-15 13:24 +0000 [c87e7dd9ec] Richard Mudgett * autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592 2017-03-07 14:13 +0000 [10fa49e327] Mark Michelson * Add rtcp-mux support This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5 2017-03-14 08:49 +0000 [dc4cdafd42] Torrey Searle * res/res_pjsip_refer: call xfer w/o extension When transfering to a URI without an extension, ensure that the s extension of the dialplan is entered ASTERISK-26869 #close Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525 2017-03-09 11:05 +0000 [982d6173c5] Sean Bright * app_queue: Handle the caller being redirected out of a queue bridge A caller can leave the Queue() application after being bridged with a member in a few ways: * Caller or member hangup * Caller is transferred somewhere else (blind or atx) * Caller is externally redirected elsewhere The first 2 scenarios are currently handled by subscribing to stasis messages, but the 3rd is not explicitly covered. If a caller is redirected away from the Queue() application, the member who was last bridged with that caller will remain in an "In use" state until the caller hangs up. This patch adds handling of the caller leaving the queue via redirection. We monitor the caller-member bridge, and if the caller is the one that leaves, we treat it the same as we would a caller hangup. ASTERISK-26400 #close Reported by: Etienne Lessard Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334 2017-03-15 08:44 +0000 [0b8a57af6d] Joshua Colp * res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. This change ensures that if no header_match option is set on an identify an error message is not output stating the option is set to an invalid value. ASTERISK-26863 Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a 2017-03-14 07:50 +0000 [1475604eff] Matt Jordan * res_pjsip_endpoint_identifier_ip: Add an option to match requests by header This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit 30f52d79d7fc9ab0b628bef2b61ea515413795a2) 2017-03-14 16:16 +0000 [f997090877] Richard Mudgett * pbx.c: Fix crash from malformed exten pattern. Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a 2017-03-14 16:51 +0000 [0dc007e94d] Richard Begg * chan_iax2: Reload of iax peer results in loss of host address/port When using a non-dynamic peer address, build_peer() invalidates the peer address structure by setting the address family to unspecified. However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() will not amend the peer address if the cache is still valid, resulting in peer connectivity failures. To fix this, we call ast_dnsmgr_refresh() instead. ASTERISK-26865 Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082 2017-03-14 15:12 +0000 [59130260e7] Matt Jordan * configure: Don't use the progress bar with curl when downloading to stdout In some scenarios, such as when there may not be a terminal (such as inside a Docker container), curl will apparently direct the progress bar to stdout. This can cause extra data to be appended to a file curl'd down to stdout, resulting in md5 verification failures. This patch removes the progress bar, and tells curl to download the file silently. ASTERISK-26872 #close Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c 2017-03-02 17:11 +0000 [8470c2bdea] George Joseph * RFC sdp: Initial SDP creation * Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48 2017-03-14 09:55 +0000 [05713c36ea] Matt Jordan * configs/samples/hep.conf.sample: Clarify how the HEP stack works This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124 2017-03-14 09:59 +0000 [0ded269bfa] Matt Jordan * funcs/func_devstate: Remove new line in Device field of during module load During module loading of func_devstate, Asterisk emits the current device state of all Custom device states currently stored in the AstDB. This was erroneously including a new line character ('\n') to the end of the device state, causing two new lines to be emitted in DeviceStateChange AMI events. Note that this only happened for those device state changes that occurred during startup. Regular device state changes for Custom device states are handled elsewhere, and did not have the newline. ASTERISK-26643 #close Reported by: Roman Bedros Tested by: Matt Jordan patches: ami_devstate.diff uploaded by Roman Bedros (License 6842) Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93 2017-03-14 09:37 +0000 [b03b72717f] Matt Jordan * main/stasis_cache: Demote the ERROR message when removing a nonexistent item This patch demotes the ERROR message that is displayed when a nonexistent item is removed from the Stasis cache. The genesis of this demotion is due to chan_sip's realtime peers and their interaction with Asterisk's core ast_endpoint code, but ostensibly it could happen from other channel drivers as well. Since Mark Michelson already did an excellent job of explaining on this issue, it is quoted here for posterity: "Internally, when a realtime peer is retrieved, Asterisk creates an ast_endpoint structure. When that peer is destroyed, the ast_endpoint is destroyed as well. Part of the destruction of the ast_endpoint involves clearing the Stasis cache of all information about that endpoint. The problem here is that the act of creating the ast_endpoint is not enough to actually put any information in the Stasis cache. Instead, something has to happen, such as a state change, in order for the Stasis cache to have any information about that endpoint. When a device registers, chan_sip creates an ast_endpoint structure, processes the REGISTER, and then destroys the ast_endpoint. When the ast_endpoint is destroyed, there is nothing to destroy in the Stasis cache, so an error message is emitted. When you use rtcachefriends, ast_endpoint structures persist for the lifetime of the module and so you do not see this error message." ASTERISK-25237 #close Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70 2017-03-08 12:39 +0000 [2d7e68c075] Matt Jordan * res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue Tabs > spaces. Always. Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1 2017-03-12 09:21 +0000 [12460b05c1] Joshua Colp * chan_pjsip: Don't assume a session will have a channel. When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01 2017-03-10 20:29 +0000 [d1ef127084] George Joseph * pjproject_bundled: Reduce the need for rebuilds Bundled pjproject should now only rebuild if one of the menuselect "Compiler Flags" options changes. Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43 2017-03-05 15:26 +0000 [36fed72614] Daniel Journo * pjsip/cli_commands: pjsip show channelstats shows wrong codec * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01 2017-03-08 14:29 +0000 [b14724adb3] Daniel Journo * res_musiconhold: moh general section is a class and issues warning * res_musiconhold.c: Ensure the general section is not treated as a moh class. ASTERISK-26353 #close Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d 2017-03-08 17:08 +0000 [35cfd2c0cc] Sean Bright * media_cache: Prefer ast_file_is_readable() over access() Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def 2017-03-07 06:25 +0000 [bc2c66b594] Sean Bright * pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel Set a variable on the channel that indicates which attempt number we are currently performing to allow for attempt-specific behavior. ASTERISK-26568 #close Reported by: Roman Shubovich Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89 2017-03-07 07:37 +0000 [4e3b0cedba] Joshua Colp * res_pjsip_transport_websocket: Add support for IPv6. This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647 2017-03-08 08:16 +0000 [60998371e3] Daniel Journo * app_voicemail: Cannot set fromstring on a per-mailbox basis * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe 2017-03-07 13:38 +0000 [5d0371d743] Mark Michelson * res_http_websocket: Fix faulty read logic. When doing some WebRTC testing, I found that the websocket would disconnect whenever I attempted to place a call into Asterisk. After looking into it, I pinpointed the problem to be due to the iostreams change being merged in. Under certain circumstances, a call to ast_iostream_read() can return a negative value. However, in this circumstance, the websocket code was treating this negative return as if it were a partial read from the websocket. The expected length would get adjusted by this negative value, resulting in the expected length being too large. This patch simply adds an if check to be sure that we are only updating the expected length of a read when the return from a read is positive. ASTERISK-26842 #close Reported by Mark Michelson Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab 2017-03-07 08:12 +0000 [d51ca4b406] Jean Aunis * chan_sip: Call not cancelled after receiving a 422 response When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099 2017-03-07 05:22 +0000 [3ed05badb9] Joshua Colp * core: Add stream topology changing primitives with tests. This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9 2017-03-06 15:54 +0000 [272259a2c6] Daniel Journo * Saynumber is trying to get "and" from "digits/" subfolder * say.c Changed 'digits/and' to 'vm-and' for en_GB ASTERISK-26598 #close Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe 2017-03-06 13:15 +0000 [5a74abc53b] Sean Bright * pbx_spool: Gracefully handle long lines in call files Per the linked issue, we aren't checking the buffer filled by fgets() to determine if it contains a newline, so we will fail to correctly parse the trailing portion of a long line. This patch increases the buffer size from 256 to 1024, and skips any line that exceeds that length, logging a warning in the process. ASTERISK-17067 #close Reported by: Dave Olszewski Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0 2017-03-02 21:27 +0000 [c9296b23d1] Richard Mudgett * core: Cleanup ast_get_hint() usage. * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b 2017-02-16 04:22 +0000 [7922f26cb0] Jørgen H * res_pjsip WebRTC/websockets: Fix usage of WS vs WSS. According to the RFC[1] WSS should only be used in the Via header for secure Websockets. * Use WSS in Via for secure transport. * Only register one transport with the WS name because it would be ambiguous. Outgoing requests may try to find the transport by name and pjproject only finds the first one registered. This may mess up unsecure websockets but the impact should be minimal. Firefox and Chrome do not support anything other than secure websockets anymore. * Added and updated some debug messages concerning websockets. * security_events.c: Relax case restriction when determining security transport type. * The res_pjsip_nat module has been updated to not touch the transport on Websocket originating messages. [1] https://tools.ietf.org/html/rfc7118 ASTERISK-26796 #close Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12 2017-02-24 15:30 +0000 [0560c32375] George Joseph * stream: Unit tests for stream read and tweaks framework * Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2 2017-03-01 07:23 +0000 [1dacf317f3] Sean Bright * res_config_pgsql: Make 'require' return consistent with other backends res_config_pgsql should match the behavior of other realtime backend drivers so that queue_log can disable adaptive logging. ASTERISK-25628 #close Reported by: Dmitry Wagin Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372 2017-02-22 15:11 +0000 [9c55a71798] Mark Michelson * SDP: Add initial SDP state machine. This introduces and documents the various states in the state machine. This also introduces API functions that induce state changes, and places TODO comments telling what needs to be done in addition to what is already there. Those TODOs will be replaced with real code in upcoming changes. Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed 2017-02-28 13:48 +0000 [60e9e4fcc0] Sean Bright * media_cache: Mark cache entry stale if cache file is removed In the event that a cache file is removed out from under us, we should treat the cache entry as stale and force a refresh. ASTERISK-26774 #close Reported by: Igor Gamayunov Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52 2017-02-28 09:41 +0000 [e5b44c26b4] Sean Bright * res_config_pgsql: Release table locks where appropriate The find_table() functions NULL or a locked table pointer. We are not consistently calling release_table() in failure paths. Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544 2017-02-28 05:41 +0000 [6ebdcfe27d] Tzafrir Cohen * pjsip.conf.sample: user_agent: not a specific version Use the description of useragent from sip.conf here. ASTERISK-26825 #close Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755 2017-02-27 20:07 +0000 [fb68db87b1] George Joseph * res_pjsip_pubsub: Remove unneeded endpoint unref When a subscription was being recreated and the endpoint wasn't found, we were trying to unref the endpoint. This was causing FRACKs. Removed the unref. ASTERISK-26823 #close Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164 2017-02-16 04:16 +0000 [ee0a123f43] Jørgen H * res_pjsip: Fix crash when contact has no status This change fixes an assumption in res_pjsip that a contact will always have a status. There is a race condition where this is not true and would crash. The status will now be unknown when this situation occurs. ASTERISK-26623 #close Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5 2017-02-21 18:06 +0000 [22242fef5d] George Joseph * res_pjsip_outbound_registration: Subscribe to network change events Outbound registration now subscribes to network change events published by res_stun_monitor and refreshes all registrations when an event happens. The 'pjsip send (un)register' CLI commands were updated to accept '*all' as an argument to operate on all registrations. The 'PJSIP(Un)Register' AMI commands were also updated to accept '*all'. ASTERISK-26808 #close Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25 2017-02-27 12:25 +0000 [4692a32ed7] George Joseph * build: Warn if asterisk is installed in both 32 and 64 bit sys dirs ... and clean them both up on uninstall. We've fixed the issue where 'make install' was installing to /usr/lib on 64-bit systems that use /usr/lib64. Now we need to clean up the remnants in /usr/lib. * 'make install' now prints a warning if DESTDIR/ASTLIBDIR contains 'lib64' and libasterisk* shared libraries or modules are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. * 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. ASTERISK-26705 Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f 2017-02-27 07:02 +0000 [ff2b4308d1] Joshua Colp * bridge_native_rtp: Handle case where channel joins already suspended. The bridge_native_rtp module did not properly handle the case where a smart bridge operation occurs while a channel is suspended. In this scenario the module would incorrectly set up local or remote RTP bridging despite the media having to flow through Asterisk. The remote endpoint would see two media streams and experience wonky audio. The module has been changed so that it ensures both channels are not suspended when performing the native RTP bridging and this requirement has been documented in the bridge technology. ASTERISK-26781 Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c 2016-08-12 11:23 +0000 [5b1796f59d] frahaase * Binaural synthesis (confbridge): DTMF conference management. DTMF configuration options for the binaural softmix bridge: toggle binaural rendering (per channel). ASTERISK-26292 Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8 2017-02-24 11:49 +0000 [2046743938] Joshua Colp * config: Improve documentation and behavior of outbound_proxy option. This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593 2017-02-23 13:03 +0000 [c07c6714f2] Joshua Colp * channel: Add ast_read_stream function for reading frames from all streams. This change introduces an ast_read_stream function and callback in the channel technology which allows reading frames from all streams and not just the default streams. The stream number has also been added to frames. This is to allow the case where frames are queued onto the channel instead of being read directly from the driver. This change does impose a restriction on reading though: a chain of frames can only contain frames from the same stream. ASTERISK-26816 Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce 2017-02-09 18:05 +0000 [a537dae6d0] George Joseph * pjproject_bundled: Update for pjproject 2.6 * Removed all 2.5.5 functional patches. * Updated usages of pj_release_pool to be "safe". * Updated configure options to disable webrtc. * Updated config_site.h to disable webrtc in pjmedia. * Added Richard Mudgett's recent resolver patches. Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7 2017-02-23 15:49 +0000 [b0067bcf2c] George Joseph * build: Execute ldconfig to build cache. (take two) On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. To make matters worse, options were being passed to ldconfig on both Linux and FreeBSD that actually prevented the rebuild of the cache. * Fedora has a /usr/share/config.site that automatically tells autoconf to use /usr/lib64 but CentOS does not. This logic was copied to configure.ac and modified so systems like Ubuntu, which still use /usr/lib for 64-bit systems, aren't affected. Now that we have them in the correct directory... In order for the system loader to find libasteriskssl and libasteriskpj, one of 3 things has to happen... - The linker cache must be rebuilt including the directory where the libasterisk* libraries were installed. Only root can rebuild the cache. This was busted. - We have to link the asterisk binary with an rpath pointing to the directrory where the libasterisk* libraries were installed. This makes things very complicated and will happen over the collective dead bodies of everyone who's had to package a distribution with an rpath. - Finally, you can start asterisk with LD_LIBRARY_PATH set to the directrory where the libasterisk* libraries were installed. There are no other options. So... * The invokation of ldconfig has been moved from main/Makefile to ASTTOPDIR/Makefile, the options have been removed, and DESTDIR/ASTLIBDIR appended. If you aren't root, you will be warned after the "Asterisk Installation Compete" banner that you must re-run 'make install' as root, manually run 'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with LD_LIBRARY_PATH. ASTERISK-26705 Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982 2017-02-23 14:48 +0000 [0f4b349d37] Sean Bright * res_config_pgsql: Fix thread safety problems * A missing AST_LIST_UNLOCK() in find_table() * The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were not consistently locking before calling it. * There were a handful of other places where pgsqlConn was accessed directly without appropriate locking. Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed 2017-02-22 05:00 +0000 [6cc890b880] Joshua Colp * channel: Add support for writing to a specific stream. This change adds an ast_write_stream function which allows writing a frame to a specific media stream. It also moves ast_write() to using this underneath by writing media frames provided to it to the default streams of the channel. Existing functionality (such as audiohooks, framehooks, etc) are limited to being applied to the default stream only. Unit tests have also been added which test the behavior of both non-multistream and multistream channels to confirm that the write() and write_stream() callbacks are invoked appropriately. ASTERISK-26793 Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c 2016-08-12 11:23 +0000 [094c26aa68] frahaase * Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix. Adds binaural synthesis to bridge_softmix (via convolution using libfftw3). Binaural synthesis is conducted at 48kHz. For a conference, only one spatial representation is rendered. The default rendering is applied for mono-capable channels. ASTERISK-26292 Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf 2017-02-22 08:53 +0000 [e57961db84] Sean Bright * res_config_ldap: Various code improvements The initial motivation for this patch was to properly handle memory allocation failures - we weren't checking the return values from the various LDAP library allocation functions. In the process, because update_ldap() and update2_ldap() were substantially the same code, they've been consolidated. Change-Id: Iebcfe404177cc6860ee5087976fe97812221b822 2017-02-22 13:08 +0000 [66a35e2451] Michael L. Young * build_tools: Fix download_externals to allow the use of curl or wget Not sure if this is really a bug versus an improvement. I can see it being viewed as a bug though by some. The current build_tools/download_externals file depends on wget in order to download external modules. The current build system is able to discover which tool to use for fetching remote files - either wget or curl. This patch takes advantage of this capability by modifying the two calls to the wget binary to instead use what was discovered by the build system. ASTERISK-26812 #close Change-Id: If9411a2554f009274d377445613ae91192d948a1 2017-02-22 11:12 +0000 [ced73d5b79] Joshua Colp * Revert "build: Execute ldconfig to build cache." This reverts commit 28c8e4f58f0f38792c7c79a05bd07788ebf15332. Change-Id: Ie2e1aaf61fd49045994974a4581545ac8348fe4c 2017-02-21 10:47 +0000 [15ed7af027] Sean Bright * pbx_realtime: Prevent premature extension matching The patterns provided by pbx_realtime were checked in the order in which they were returned from the realtime backend. If there was overlap between multiple patterns, the first one to correctly match was chosen even though it may not have been the best match. We now sort the patterns descending by their length and compare in that order. There may be cases where this still results in a sub-optimal match, but this patch should improve the overall behavior. ASTERISK-18271 #close Reported by: Charlie Smurthwaite Change-Id: I56d9ac15810eb1775966b669c3028e32cc7bd809 2017-02-22 08:32 +0000 [f58aefba5b] Joshua Colp * core: Show streams in "core show channel". The "core show channel" CLI command will now output the streams present on the channel with their details. ASTERISK-26811 Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a 2017-02-21 15:09 +0000 [fc70ca9499] Sean Bright * pbx_dundi: DUNDi weight parameter not processed correctly The DUNDi weight field is not always converted from network byte order to host byte order. This can result in incorrect weight values and incorrect selection of DUNDi destinations. ASTERISK-18731 #close Reported by: Peter Racz Patches: dundi_weight.patch (license #6290) patch uploaded by Peter Racz Change-Id: Iba3e1a700ff539db57211a7bbc26f7b22ea9a1be 2017-02-15 14:43 +0000 [a738772edd] Mark Michelson * Add initial SDP state code. This establishes the basic allocation/destruction of an SDP state object, plus some of the simpler getter methods involved. Subsequent tasks will deal with adding a state machine, creating SDPs from capabilities and options, and merging SDPs into a joint SDP. Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709 2017-02-21 10:47 +0000 [ab04a018e4] Sean Bright * realtime: Fix ast_load_realtime_multientry handling ast_load_realtime_multientry() returns an ast_config structure whose ast_categorys are keyed with the empty strings. Several modules were giving semantic meaning to the category names causing problems at runtime. * app_directory: Treated the category name as the mailbox name, and would fail to direct calls to the appropriate extension after an entry was chosen. * app_queue: Queues, queue members, and queue rules were all affected and needed to be updated. * pbx_realtime: Pattern matching would never succeed because the extension entered by the user was always compared to the empty string. Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7 2017-02-21 08:56 +0000 [6e6c96d713] Sean Bright * realtime: Centralize some common realtime backend code All of the realtime backends create artificial ast_categorys to pass back into the core as query results. These categories have no filename or line number information associated with them and the backends differ slightly on how they create them. So create a couple helper macros to help make things more consistent. Also updated the call sites to remove redundant error messages about memory allocation failure. Note that res_config_ldap sets the category filename to the 'table name' but that is not read by anything in the core, so I've dropped it. Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897 2017-02-16 10:30 +0000 [28c8e4f58f] Joshua Colp * build: Execute ldconfig to build cache. On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. If DESTDIR is specified, however, the old logic is executed as the install process may not have permission to alter the ldconfig cache. ASTERISK-26705 Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4 2017-01-08 20:32 +0000 [6f15500ced] Richard Mudgett * res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation. The inbound authentication object is supposed to be immutable when it is stored in sorcery. However, the immutable property is violated if the authentication object does not have a realm set. The immutable contract violation has a different effect depending upon what sorcery back end is used. If it is the config file back end you would get the same object back until res_pjsip is reloaded. If it is the real-time or AstDB back end you would get a new object on each query. If it is cached you would get the same object back until it is refreshed from the database. Once an inbound authentication object has its realm set it may or may not get updated again if the default_realm changes. If the same authentication object is used for inbound and outbound authentication then the immutable violation can make it very hard to determine why the outbound authentication now fails. The only diagnostic message is a complaint about no realms matching when it had worked earlier. It fails because of the difference in behaviour for an empty realm setting between inbound and outbound authentication objects. * Fixed the sorcery object immutable violation by creating a new object and setting the default_realm on it instead. The new object is a shallow copy for speed. * The auth_store thread storage no longer holds an auth ref. It interferes with the shallow copy and never needed a ref anyway. ASTERISK-26799 #close Change-Id: I2328a52f61b78ed5fbba38180b7f183ee7e08956 2017-02-04 20:17 +0000 [6400f5f309] Richard Mudgett * res_pjsip: Update artificial auth whenever default_realm changes. There was code attempting to update the artificial authentication object whenever the default_realm changed. However, once the artificial authentication object was created it would never get updated. The artificial authentication object would require a system restart for a change to the default_realm to take effect. ASTERISK-26799 Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802 2017-01-01 08:02 +0000 [0b660c9989] Richard Mudgett * res_pjsip: Update authentication realm documentation. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. An empty inbound auth realm represents the global section's default_realm value when the authentication object is used to challenge an incoming request. An empty outgoing auth realm is treated as a don't care wildcard when the authentication object is used to respond to an incoming authentication challenge. ASTERISK-26799 Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce 2017-02-13 17:11 +0000 [7f83bcd63d] Richard Mudgett * pjproject: Fixes to resolve DNS SRV crashes. * Re #1945 (misc): Don't trigger SRV complete callback when there is a parse error. * srv_resolver.c: Don't try to send query if already considered resolved. ** In resolve_hostnames() don't try to resolve a query that is already considered resolved. ** In resolve_hostnames() fix DNS typo in comments. ** In build_server_entries() move a common expression assigning to cnt earlier. * sip_transport.c: Fix tdata object name to actually contain the pointer. It helps if the logs referencing a tdata object buffer actually have a name that includes the correct pointer as part of the name. Also since the tdata has its own pool it helps if any logs referencing the pool have the same name as the tdata object. This change brings tdata logging in line with how tsx objects are named. ASTERISK-26669 #close ASTERISK-26738 #close Change-Id: I56af2ded25476b3e870ca586ee69ed6954ef75af 2017-02-20 13:38 +0000 [bf78c3c9c3] Richard Mudgett * pjproject: Increase SENDER_WIDTH column size for 64-bit system logs. ASTERISK-26669 ASTERISK-26738 Change-Id: Ibae6fc8cae69a1f04df0c577c4c11200499d6fe0 2017-02-06 14:26 +0000 [54812f18b5] Richard Mudgett * pjsip_distributor.c: Update some debug messages to get transaction name. * Removed overloaded unmatched response ignore. We obviously sent the request so we shouldn't ignore it because it isn't new work. ASTERISK-26669 ASTERISK-26738 Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37 2017-02-20 06:28 +0000 [b18f1bfb13] Sean Bright * app_voicemail: vm_authenticate accesses uninitialized memory vm_authenticate doesn't always set the passed ast_vm_user argument, so we initialize to 0 before passing it in. ASTERISK-25893 #close Reported by: Filip Jenicek Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a 2017-02-20 11:19 +0000 [7739b0b3ae] Joshua Colp * Revert "build: Execute ldconfig to build cache." This reverts commit 8851c3e0885cb704a5a6159a51768ea5297e9b10. Change-Id: I124380be5e3bd57da978428a2a93604336ccd0db 2017-02-20 08:04 +0000 [ffa7d69766] George Joseph * pjproject cli: Add object count after object lists When listing a container, we now print the number of objects in the container at the end of the list. Change-Id: I791cbc3ee9da9a2af9adc655164b5d32953df812 2017-02-20 05:53 +0000 [e84353b8a8] Sean Bright * res_config_ldap: Don't try to delete non-existent attributes OpenLDAP will raise an error when we try to delete an LDAP attribute that doesn't exist. We need to filter out LDAP_MOD_DELETE requests based on which attributes the current LDAP entry actually has. There is of course a small window of opportunity for this to still fail, but it is much less likely now. Change-Id: I3fe1b04472733e43151563aaf9f8b49980273e6b 2017-02-20 05:49 +0000 [9f392574f9] Sean Bright * res_config_ldap: Remove extraneous line numbers from log messages Extraneous line numbers were being output in many log messages. These have been removed. Change-Id: Ice9efa3d252ee87f37fa8f5ea852fda482675431 2017-02-20 05:45 +0000 [ef0944395e] Sean Bright * res_config_ldap: Make memory allocation more consistent The code in update_ldap() and update2_ldap() was using both Asterisk's memory allocation routines as well as OpenLDAP's. I've changed it so that everything that is passed to OpenLDAP's functions are allocated with their routines. Change-Id: Iafec9c1fd8ea49ccc496d6316769a6a426daa804 2017-02-20 05:30 +0000 [dd3efdf525] Sean Bright * res_config_ldap: Fix configuration inheritance from _general The "_general" configuration section allows administrators to provide both general configuration options (host, port, url, etc.) as well as a global realtime-to-LDAP-attribute mapping that is a fallback if one of the later sections do not override it. This neglected to exclude the general configuration options from the mapping. As an example, during my testing, chan_sip requested 'port' from realtime, and because I did not have it defined, it pulled in the 'port' configuration option from "_general." We now filter those out explicitly. Change-Id: I1fc61560bf96b8ba623063cfb7e0a49c4690d778 2017-02-20 05:27 +0000 [d6d86f1c09] Sean Bright * res_config_ldap: Fix erroneous LDAP_MOD_REPLACE in LDAP modify We always treat the first change of our modification batch as a replacement when it sometimes is actually a delete. So we have to pass the correct arguments to the OpenLDAP library. ASTERISK-26580 #close Reported by: Nicholas John Koch Patches: res_config_ldap.c-11.24.1.patch (license #6833) patch uploaded by Nicholas John Koch Change-Id: I0741d25de07c9539f1edc6eff3696165dfb64fbe 2017-02-15 11:55 +0000 [44abe214d2] Sean Bright * res_config_sqlite3: Fix crash when loading with invalid config When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has already destroyed the ast_config struct for us. Trying to do it again results in a crash. Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6 2017-02-17 17:06 +0000 [51e3b11989] Sean Bright * pjproject-bundled: Fix checksum verification when using cURL ASTERISK-26802 #close Reported by: Michael L. Young Change-Id: Iad293080f55d4d69ab615717a15211d916eed613 2017-02-17 16:57 +0000 [0b427f9b59] Richard Mudgett * tcptls.c: Add some missing allocation failure checks. * Fix tcptls_session ref and fd leak in ast_tcptls_server_root(). Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb 2017-02-17 14:58 +0000 [dbc3598014] Mark Michelson * Remove extra ast_iostream_close() calls. When AMI encounters an error at the beginning of a session, it would explicitly call ast_iostream_close() on its tcptls session's iostream. It then would jump to a label where it would shut down the tcptls session instance. The tcptls session instance would again attempt to close the iostream. Under normal circumstances, this might go by unnoticed. However, when MALLOC_DEBUG is enabled, all fields on the iostream get set to 0xdeaddead when the iostream is freed. Thus a second call to ast_iostream_close() after the iostream has been freed would reslt in an attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn horribly. The fix here is to not directly close the iostream from the dangerous scenarios. The specific scenarios are: * Exceeding the configured authlimit * Failing to build a mansession on a new connection Change-Id: I908f98d516afd5a263bd36b072221008a4731acd 2017-02-14 09:54 +0000 [5a130b2e17] Mark Michelson * Add SDP translator and PJMEDIA implementation. This creates the following: * Asterisk's internal representation of an SDP * An API for translating SDPs from one format to another * An implementation of a translator for PJMEDIA Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b 2017-02-07 09:50 +0000 [8af6342555] Mark Michelson * Add initial SDP options. This is step one of adding an SDP API: defining some configurable settings for SDPs. This is based on options that are currently supported in Asterisk. Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7 2017-02-16 10:30 +0000 [8851c3e088] Joshua Colp * build: Execute ldconfig to build cache. On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. ASTERISK-26705 Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519 2017-02-16 08:38 +0000 [e93f2a5142] Sean Bright * realtime: Fix LIKE escaping in SQL backends The realtime framework allows for components to look up values using a LIKE clause with similar syntax to SQL's. pbx_realtime uses this functionality to search for pattern matching extensions that start with an underscore (_). When passing an underscore to SQL's LIKE clause, it will be interpreted as a wildcard matching a single character and therefore needs to be escaped. It is (for better or for worse) the responsibility of the component that is querying realtime to escape it with a backslash before passing it in. Some RDBMs support escape characters by default, but the SQL92 standard explicitly says that there are no escape characters unless they are specified with an ESCAPE clause, e.g. SELECT * FROM table WHERE column LIKE '\_%' ESCAPE '\' This patch instructs 3 backends - res_config_mysql, res_config_pgsql, and res_config_sqlite3 - to use the ESCAPE clause where appropriate. Looking through documentation and source tarballs, I was able to determine that the ESCAPE clause is supported in: MySQL 5.0.15 (released 2005-10-22 - earliest version available from archives) PostgreSQL 7.1 (released 2001-04-13) SQLite 3.1.0 (released 2005-01-21) The versions of the relevant libraries that we depend on to access MySQL and PostgreSQL will not work on versions that old, and I've added an explicit check in res_config_sqlite3 to only use the ESCAPE clause when we have a sufficiently new version of SQLite3. res_config_odbc already handles the escape characters appropriately, so no changes were required there. ASTERISK-15858 #close Reported by: Humberto Figuera ASTERISK-26057 #close Reported by: Stepan Change-Id: I93117fbb874189ae819f4a31222df7c82cd20efa 2017-02-16 08:28 +0000 [f8f513d363] George Joseph * stream: Rename creates/destroys to allocs/frees To be consistent with sdp implementation. Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500 2017-02-16 05:46 +0000 [30aaeec5a1] Sean Bright * res_config_sqlite3: Properly create missing columns when necessary There were two specific issues resolved here: 1) The code that iterated over the required fields (via ast_realtime_require) was broken for the RQ_INTEGER1 field type. Iteration would stop when the first RQ_INTEGER1 (0) field was encountered. 2) sqlite3_changes() was used to try and count the number of rows returned by a SELECT statement. sqlite3_changes() only counts affected rows, so this was always returning the value from the most recent data modification statement. We now separate read-only queries from data modification queries and count rows appropriately in both cases. ASTERISK-23457 #close Reported by: Scott Griepentrog Change-Id: I91ed20494efc3fcfbc2a96ac7646999a49814884 2017-02-15 14:44 +0000 [ac7a34c531] Joshua Elson * http: Ensure capath is defined on all http creations ASTERISK-26794 #close Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1 2017-02-15 23:09 +0000 [135bea931c] Igor Goncharovsky * chan_unistim: fix char type to have consistent behavior on ARM There is difference exists in behaviour of char type on x86 and ARM. On x86 by default char variable type means signed char, but in ARM unsigned char used. This make binary calculations and negative values works wrong on ARM. This patch change type of char variables used for store negative values and binary calculations to signed char. ASTERISK-26714 Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab 2017-02-07 13:17 +0000 [4bdf5d329f] George Joseph * res_pjsip_pubsub: Correctly implement persisted subscriptions This patch fixes 2 original issues and more that those 2 exposed. * When we send a NOTIFY, and the client either doesn't respond or responds with a non OK, pjproject only calls our pubsub_on_evsub_state callback, no others. Since pubsub_on_evsub_state (which does the sub_tree cleanup) does not expect to be called back without the other callbacks being called first, it just returns leaving the sub_tree orphaned. Now pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE which is what pjproject will set to tell us that it was the transaction that timed out or failed and not the subscription itself timing our or being terminated by the client. If is TSX_STATE, pubsub_on_evsub_state now does the proper cleanup regardless of the state of the subscription. * When a client renews a subscription, we don't update the persisted subscription with the new expires timestamp. This causes subscription_persistence_recreate to prune the subscription if/when asterisk restarts. Now, pubsub_on_rx_refresh calls subscription_persistence_update to apply the new expires timestamp. This exposed other issues however... * When creating a dialog from rdata (which sub_persistence_recreate does from the packet buffer) there must NOT be a tag on the To header (which there will be when a client refreshes a subscription). If there is one, pjsip_dlg_create_uas will fail. To address this, subscription_persistence_update now accepts a flag that indicates that the original packet buffer must not be updated. New subscribes don't set the flag and renews do. This makes sure that when the rdata is recreated on asterisk startup, it's done from the original subscribe packet which won't have the tag on To. * When creating a dialog from rdata, we were setting the dialog's remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. When the client tried to resubscribe after a restart with the correct cseq, we'd reject the request with an Invalid CSeq error. * The acts of creating a dialog and evsub by themselves when recreating a subscription does NOT restart pjproject's subscription timer. The result was that even if we did correctly recreate the subscription, we never removed it if the client happened to go away or send a non-OK response to a NOTIFY. However, there is no pjproject function exposed to just set the timer on an evsub that wasn't created by an incoming subscribe request. To address this, we create our own timer using ast_sip_schedule_task. This timer is used only for re-establishing subscriptions after a restart. An earlier approach was to add support for setting pjproject's timer (via a pjproject patch) and while that patch is still included here, we don't use that call at the moment. While addressing these issues, additional debugging was added and some existing messages made more useful. A few formatting changes were also made to 'pjsip show scheduled tasks' to make displaying the subscription timers a little more friendly. ASTERISK-26696 ASTERISK-26756 Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e 2017-02-15 11:03 +0000 [11886dea82] Sean Bright * res_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16 pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND, which is a compile-time constant. Instead of hard-coding 16 when we enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can potentially collect more interfaces if the compile time options are changed. Tangentially related to ASTERISK~24464 Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd 2016-12-22 09:42 +0000 [b58de2fab7] Dennis Guse * Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer Adds the import tool for converting a HRIR database to hrirs.h ASTERISK-26292 Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547 2017-02-14 12:33 +0000 [a9c15a0e4c] Joshua Colp * stream: Add unit tests for channel stream usage. This change adds unit tests cover the following: 1. That retrieving the first media stream of a specific media type from a stream topology retrieves the expected media stream. 2. That setting the native formats of a channel which does not support streams results in the creation of streams on its behalf according to the formats of the channel. 3. That setting a stream topology on a channel which supports streams sets the topology to the provided one. ASTERISK-26790 Change-Id: Ic53176dd3e4532e8c3e97d9e22f8a4b66a2bb755 2017-02-13 16:50 +0000 [275f469a4d] Sean Bright * app_voicemail: Allow 'Comedian Mail' branding to be overriden Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6 2017-02-13 11:50 +0000 [bf2f091bbb] George Joseph * stream: Add stream topology to channel Adds topology set and get to channel. ASTERISK-26790 Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4 2017-01-25 16:25 +0000 [2b245b12d9] Ryan Rittgarn * app_voicemail: VoiceMailPlayMsg did not play database stored messages When attempting to use VoiceMailPlayMsg with a realtime data backend the message is located, but never retrieved. This patch adds the required RETRIEVE and DISPOSE calls that will fetch the message from the database (and IMAP storage as well for that matter). Also, removed extraneous make_file call. ASTERISK-26723 #close Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c 2017-02-14 08:12 +0000 [662c9e69fa] Sean Bright * app_record: Add option to prevent silence from being truncated When using Record() with the silence detection feature, the stream is written out to the given file. However, if only 'silence' is detected, this file is then truncated to the first second of the recording. This patch adds the 'u' option to Record() to override that behavior. ASTERISK-18286 #close Reported by: var Patches: app_record-1.8.7.1.diff (license #6184) patch uploaded by var Change-Id: Ia1cd163483235efe2db05e52f39054288553b957 2017-02-07 11:13 +0000 [9f394d074a] Sebastian Gutierrez * app_queue: reset abandoned in sl for sl2 calculations ASTERISK-26775 #close Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284 2017-02-13 11:00 +0000 [6c4657e28e] Joshua Colp * stream: Add stream topology unit tests and fix uncovered bugs. This change adds unit tests for the various API calls relating to stream topologies. This includes creation, destruction, inspection, and manipulation. Through this a few bugs were uncovered in the implementation: 1. Creating a topology using a format capabilities would fail as the code considered a return value of 0 from the append stream function to indicate an error which is incorrect. 2. Not all functions which placed a stream into a topology set the position on the stream itself. 3. Appending a stream would cause a frack if the position provided was the last one. This occurred because the existing stream was queried but the index was outside of what the vector was currently at for size. ASTERISK-26786 Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0 2017-02-11 09:57 +0000 [3f94373778] Sean Bright * cli: Fix various CLI documentation and completion issues * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca 2017-02-10 15:45 +0000 [8b72ec312b] George Joseph * stream: Add media stream topology definition and API This change adds the media stream topology definition and API for accessing and using it. Some refactoring of the stream was also done. ASTERISK-26786 Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568 2017-01-13 11:21 +0000 [75f8167e66] Norbert Varga * chan_pjsip: Multidomain endpoint finding on call When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), the user part is stripped down as it would be a trunk with a specified user, and only the host part is called as a PJSIP endpoint and can't be found. This is not correct in the case of a multidomain SIP account, so the stripping after the @ sign is done only if the whole endpoint (in multidomain case 1000@test.com) can't be found. ASTERISK-26248 Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6 2017-02-13 05:05 +0000 [89871576b9] Joshua Colp * channel: Protect flags in ast_waitfor_nandfds operation. The ast_waitfor_nandfds operation will manipulate the flags of channels passed in. This was previously done without the channel lock being held. This could result in incorrect values existing for the flags if another thread manipulated the flags at the same time. This change locks the channel during flag manipulation. ASTERISK-26788 Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed 2017-02-11 11:25 +0000 [07abb39d6a] Richard Mudgett * res_pjsip.c: Fix inconsistency between warning and action. The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE but we have no authenticator registered to create the challenge. Change-Id: I62368180d774b497411b80fbaabd0c80841f8512 2017-02-11 11:26 +0000 [ce810a892b] Richard Mudgett * pjsip_distributor.c: Fix off-nominal tdata ref leak. Change-Id: I571f371d0956a8039b197b4dbd8af6b18843598d 2017-02-09 10:01 +0000 [0910773077] Sean Bright * manager: Restore Originate failure behavior from Asterisk 11 In Asterisk 11, if the 'Originate' AMI command failed to connect the provided Channel while in extension mode, a 'failed' extension would be looked up and run. This was, I believe, unintentionally removed in 51b6c49. This patch restores that behavior. This also adds an enum for the various 'synchronous' modes in an attempt to make them meaningful. ASTERISK-26115 #close Reported by: Nasir Iqbal Change-Id: I8afbd06725e99610e02adb529137d4800c05345d 2017-02-08 14:27 +0000 [16fdb11bc3] Richard Mudgett * core: Cleanup some channel snapshot staging anomalies. We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a 2017-02-07 06:56 +0000 [bab4885f1e] Joshua Colp * stream: Add media stream definition and API with unit tests. This change adds the media stream definition and API for accessing and using it. Unit tests have also been written which exercise aspects of the API. ASTERISK-26773 Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87 2017-02-10 09:35 +0000 [648d181d2f] George Joseph * configs/samples: Fix placement of 'identify' entry in sorcery.conf The entry for 'identify' was incorrectly placed in the res_pjsip section when it should be in res_pjsip_endpoint_identifier_ip. ASTERISK-26785 #close Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a 2017-02-08 11:50 +0000 [46147a8f30] Mark Michelson * Revert "Update qualifies when AOR configuration changes." This reverts commit 6492e91392b8fd394193e411c6eb64b45486093f. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e 2017-02-07 12:01 +0000 [5422ec140c] Joshua Colp * srv: Fix crash when ast_srv_lookup is used and 0 records are returned. When performing an SRV lookup using the ast_srv_lookup function it did not properly handle the situation where 0 records are returned. If this happened it would wrongly assume that at least one record was present. This change fixes the code so it will exit early if an error occurs or if 0 records are returned. ASTERISK-26772 patches: srv_lookup.patch submitted by nappsoft (license 6822) Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351 2017-02-06 11:40 +0000 [b79cc62057] Joshua Colp * res_stasis_device_state: Protect the adding/removing of subscriptions. The adding and removing of device state subscriptions did not protect fully against simultaneous manipulation. In particular the subscribe case allowed a small window where two subscriptions could be added for the same device state instead of just one. This change makes the code hold the subscriptions lock for the entirety of each operation to ensure that two are not occurring at the same time. ASTERISK-26770 Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b 2017-02-01 17:56 +0000 [b47cf1a7d6] Richard Mudgett * res_pjsip: Fix some off nominal tdata leaks. Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d 2017-02-03 15:26 +0000 [7b280e7ccf] Sebastien Duthil * res_ari: fix memory leak for channelvars In ari.conf, when setting the option channelvars, every Stasis channel snapshot would create a list of variable/value that would not be freed when the snapshot is freed, resulting in a often-recurring memory leak. ASTERISK-26767 #close Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d 2017-02-03 02:25 +0000 [c6c7f17206] Tzafrir Cohen * libasteriskssl: do nothing with OpenSSL >= 1.1 OpenSSL 1.1 requires no explicit initialization. The hacks in the library are not needed. They also happen to fail running Asterisk. Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100 2017-01-20 23:59 +0000 [bc041ca14a] Tzafrir Cohen * tcptls: use TLS_client_method with OpenSSL 1.1 OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous version-specific methods (such as TLSv1_client_method(). Other than being simpler to use and more correct (gain support for TLS newer that TLS1, in our case), the older ones produce a deprecation warning that fails the build in dev-mode. Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07 2017-01-20 23:57 +0000 [2c8d0764de] Tzafrir Cohen * openssl 1.1 support: use OPENSSL_VERSION_NUMBER Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect the openssl 1.1 API. Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458 2017-01-31 18:28 +0000 [50029f585e] Richard Mudgett * channel.c: Fix unbalanced read queue deadlocking local channels. Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 2017-01-31 16:38 +0000 [97c308471d] Richard Mudgett * res_agi: Prevent an AGI from eating frames it should not. (Re-do) A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 2017-01-31 16:32 +0000 [72e3fc5845] Richard Mudgett * Frame deferral: Revert API refactoring. There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 2017-02-02 11:26 +0000 [4c51ad158d] Sean Bright * res_odbc: Remove deprecated settings from sample configuration file ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f 2017-02-01 17:14 +0000 [7d9b50a7b2] Richard Mudgett * res_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap. ASTERISK-26765 Change-Id: I27eb97df7f8d7e624b0b9a61c0fcee4718c86d8d 2017-02-01 15:56 +0000 [2849b726b6] Sean Bright * audiohooks: Muting a hook can mute underlying frames If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed 2017-02-01 13:54 +0000 [bbed75c3ba] Mark Michelson * Update qualifies when AOR configuration changes. Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121 2017-01-31 11:17 +0000 [aeea634bc0] Joshua Colp * res_pjsip: Handle invocation of callback on outgoing request when error occurs. There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91 2017-01-30 09:02 +0000 [7a16524a83] Sean Bright * res_rtp_asterisk: Swap byte-order when sending signed linear Before Asterisk 13, signed linear was converted into network byte order by a smoother before being sent over the network. We restore this behavior by forcing the creation of a smoother when slinear is in use and setting the appropriate flags so that the byte order conversion is always done. ASTERISK-24858 #close Reported-by: Frankie Chin Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9 2017-01-31 12:46 +0000 [e252aff9ad] George Joseph * debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c 2017-01-25 06:50 +0000 [ef4deb8ecd] George Joseph * debug_utilities: Add ast_logescalator The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543 2017-01-23 09:35 +0000 [178b90af02] Torrey Searle * libastssl/pj: libastssl/pj should have an so_version Issue introduced in b59956a87. In the non-darwin case libastssl/pj should be versioned. This causes the symbol file for this lib to not be generated. Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c (cherry picked from commit 54b027916a71f2b83b2050cef5ef704ea5de39b2) 2017-01-24 19:51 +0000 [138cd8d019] Kirill Katsnelson * make_build_h: handle backslashes in external strings LikewiseOpen creates user names with a backslash in them. A gentle massage with sed(1) allows such strings to be inserted into build.h properly quoted. I am also adding the same for host name and other strings used in the script that are more or less user-controlled. ASTERISK-26754 Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae 2017-01-24 22:31 +0000 [8270d2436d] Kirill Katsnelson * app_queue: Fix queues randomly disappearing on reload With 500+ queues and a reload every minute, a random queue disappears upon reload. The cause is mususe of the 'dead' flag. Namely, all queues were marked dead up front, and then "resurrected" by dropping this flag for those found in the configuration. But a queue marked dead can be removed also when control leaves the app entry point on a PBX thread. With this change, the queue is marked only not found, and at the end of reload only the queues that are still not found are actually marked as dead, so the dead flag is never reset, and set only on positively dead queues. ASTERISK-26755 Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf 2017-01-26 07:57 +0000 [7fa3de7ae9] Joshua Colp * res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving. This change adds a missing unreference of the hostname when resolving and also cleans up the iterator. ASTERISK-26735 Change-Id: Ic012ebaf3d89e714eec340b7b0c5e63c66af857a 2017-01-25 15:26 +0000 [d32bd63860] Mark Michelson * Add reload options to CLI/AMI stale object commands. Marking an object as stale in a memory cache is supposed to prime the cache so that the next time the item is retrieved, the stale item is deleted from the cache and a background task is run to re-populate the cache with a fresh version of the object. The problem is, there are some object types out there for which there is no natural reason that they would be retrieved from the backend with any regularity. Outbound PJSIP registrations are a good example of this. At startup, they are read, and an object-specific state is created that refers to the initially-retrieved object for all time. Adding the "reload" option to the CLI/AMI commands gives the cache the opportunity to manually re-retrieve the object from the backend, both storing the new object in the cache and applying the new object's configuration to the module that uses that object. Change-Id: Ieb1fe7270ceed491f057ec5cbf0e097bde96c5c8 2017-01-10 17:39 +0000 [20aed30d9a] Richard Mudgett * T.140: Fix format ref and memory leaks. * channel.c:ast_sendtext(): Fix T.140 SendText memory leak. * format_compatibility.c: T.140 RED and T.140 were swapped. * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic scheduled red_write(). * res_rtp_asterisk.c: Some other minor misc tweaks. Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb 2017-01-24 15:39 +0000 [ee2b0f2eef] Joshua Colp * res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0. When configuring a match using a netmask the error variable was not defaulting to 0. For some people this would cause the code to think an error occurred when adding the match when in reality it added perfectly fine. ASTERISK-26693 Change-Id: I850c250813742bddde65c84e739093c9e01dfe56 2017-01-10 17:37 +0000 [930a24a730] Richard Mudgett * astobj2.c: Add excessive ref count trap. Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a 2017-01-10 13:11 +0000 [de28c1b9f1] Richard Mudgett * main/app.c: Memory corruption from early format destruction. * make_silence() created a malloced silence slin frame without adding a slin format ref. When the frame is destroyed it will unref the slin format that never had a ref added. Memory corruption is expected to follow. * Simplified and fixed counting the number of samples in a frame list for make_silence(). * Eliminated an unnecessary RAII_VAR associated with the make_silence() frame. Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747 2017-01-11 14:59 +0000 [2039eb8edf] Richard Mudgett * frame.c: Fix off-nominal format ref leaks. * ast_frisolate() could leak frame format refs on allocation failures. * Similified code in ast_frisolate() and code used by ast_frisolate(). Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d 2017-01-13 19:08 +0000 [e922979d49] Richard Mudgett * stasis_bridge.c: Fix off-nominal stasis control ref leak. Change-Id: Ib17218343a6596832060180e19386da9df150ac8 2017-01-10 12:30 +0000 [56854f22d2] Richard Mudgett * res_musiconhold.c: Fix format ref leak when parsing MOH config class. Change-Id: Ica8e8e2ce7604c2c61ec55bef07dc675361d2ea5 2017-01-10 14:03 +0000 [d87f81ddb1] Richard Mudgett * chan_oss.c: Fix format ref leak in oss_read(). Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0 2017-01-10 17:48 +0000 [36bdd7c1a0] Richard Mudgett * Add notes about embedded ast_frame structs holding a format ref. mod_format.h: Note ast_filestream.fr holds a format ref. translate.h: Note ast_trans_pvt.f holds a format ref. Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749 2017-01-20 21:13 +0000 [6f3e8c8e01] Richard Mudgett * PJPROJECT logging: Fix detection of max supported log level. The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4 2017-01-05 13:21 +0000 [0ea3c371c5] Richard Mudgett * res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands. ASTERISK-23828 #close Change-Id: Ifb8a3b61f447aedc58a8e6b36a810f7566018567 2017-01-23 16:18 +0000 [4bfeda6ee4] Mark Michelson * Free endpoint ACLs when destroying PJSIP endpoints. If endpoint ACLs were specified, they were not being freed when endpoints were destroyed. On systems with realtime endpoints, this could add up quickly since each DB lookup would allocate the ACL without freeing it. ASTERISK-26731 #close Reported by Ustinov Artem Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad 2017-01-19 09:05 +0000 [6691606723] George Joseph * ari: Implement 'debug all' and request/response logging The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f39a5a81b20da44358143ed9b54ab0fe) 2017-01-20 23:41 +0000 [f3f9175df0] Tzafrir Cohen * test_voicemail_api: order of params to VERIFY macros Fix order of parameters in calls to VM_API_INT_VERIFY and VM_API_STRING_VERIFY ASTERISK-26739 #close Change-Id: I30dc6b36893aadad6012be3f16f93aa5720870d6 Note: status: builds. Not tested any further. 2017-01-23 09:10 +0000 [96e7291cbd] George Joseph * pjproject_bundled: Fix setting max log level An earlier attempt to prevent pjsua from spitting out an extra 6795 lines of debug output every time the testsuite called it was also turning off the ability for asterisk to output debug info when it needed to. This patch reverts the earlier fix and instead adds a pjproject patch that sets the startup log level to 1 for pjsua pjsystest and the pjsua python binding. This is an asterisk-only patch that does not affect pjproject functionality and will not be submitted upstream. Change-Id: I347a8b58b2626f2906ccfc1d339e907627a0c9e8 2017-01-23 10:08 +0000 [23690c1b35] Joshua Colp * res_pjsip_endpoint_identifier_ip: Read settings before resolving. An option has been added, srv_lookups, which controls whether SRV lookups are performed on the provided match hosts or not. It was possible for this option to be applied after resolution had already happened. This change makes it so hosts are stored away, settings are read and applied, and then resolution is done. This ensures that no matter the ordering the srv_lookups option is in effect. ASTERISK-26735 Change-Id: I750378cb277be0140f8c5539450270afbfc43388 2016-11-29 09:31 +0000 [1061539b75] Lorenzo Miniero * media: Add experimental support for RTCP feedback. This change adds experimental support for providing RTCP feedback information to codec modules so they can dynamically change themselves based on conditions. ASTERISK-26584 Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857 2017-01-22 17:25 +0000 [cfe72c39cf] Richard Mudgett * LISTFILTER: Remove outdated ERROR message. Feeding LISTFILTER an empty variable results in an invalid ERROR message. Earlier changes made the message useless because we can no longer tell if the variable is empty or does not exist. It is valid to try to remove a value from an empty list just as it is valid to try to remove a value that is not in a non-empty list. * Removed the outdated ERROR message. * Added more test cases to the LISTFILTER unit test. Change-Id: Ided9040e6359c44a335ef54e02ef5950a1863134 2017-01-21 14:43 +0000 [dbb9c8141d] Tzafrir Cohen * tests: use datadir for sound files Some (voicemail-related) tests API symlinks beep.gsm and other files from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR. ASTERISK-26740 #close Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89 2017-01-05 15:11 +0000 [ef9164b9ca] Richard Mudgett * res_pjsip_pubsub.c: Fix AMI event list counts. Fix the AMI PJSIPShowSubscriptionsInbound, PJSIPShowSubscriptionsOutbound, and PJSIPShowResourceLists actions event counts. The reported counts may not necessarily be accurate depending on what happens. The subscriptions count would be wrong if Asterisk ever has outbound subscriptions. The resource list count could be wrong if a list were added or removed during the AMI action being processed. Change-Id: I4344301827523fa174960a42c413fd19abe4aed5 2017-01-05 13:02 +0000 [ab858295a2] Richard Mudgett * res_pjsip_pubsub.c: Fix incorrect message string wrapping. Change-Id: Id771e6fe56d89ce365ddcbb423f820af97211120 2017-01-05 13:01 +0000 [6d648185bc] Richard Mudgett * res_pjsip_pubsub.c: Eliminate trivial SCOPED_LOCK usage. Change-Id: Ie0b69a830385452042fa19e7d267c6790ec6b6be 2017-01-05 12:58 +0000 [90f3b1270c] Richard Mudgett * res_pjsip: alloca can never fail. Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1 2017-01-13 11:03 +0000 [d16b3a9917] George Joseph * debug_utilities: Create ast_loggrabber ast_loggrabber gathers log files from customizable search patterns, optionally converts POSIX timestamps to a readable format and tarballs the results. Also a few tweaks were made to ast_coredumper. Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 (cherry picked from commit c70915287837704090d75f181525765de7a17221) 2017-01-01 03:47 +0000 [48730ae65e] Richard Mudgett * res_pjsip_outbound_authenticator_digest.c: Fix spacing in warning messages. Change-Id: I573f0343c0c63a785cd4da60d57cc9f8b9ce7f49 2016-12-22 04:07 +0000 [40b9766a31] Martin Tomec * app_queue: add RINGCANCELED log event on caller hang up QueueLog did not log ringnoanswer when the caller abandoned call before first timeout. It was impossible to get agent membername and ringing duration for this short calls. After some discusions it seems that the best way is to add new event RINGCANCELED, which is generated after caller hangup during ringing. ASTERISK-26665 Change-Id: Ic70f7b0f32fc95c9378e5bcf63865519014805d3 2017-01-12 15:58 +0000 [283c16c6b6] Kevin Harwell * abstract/fixed/adpative jitter buffer: disallow frame re-inserts It was possible for a frame to be re-inserted into a jitter buffer after it had been removed from it. A case when this happened was if a frame was read out of the jitterbuffer, passed to the translation core, and then multiple frames were returned from said translation core. Upon multiple frames being returned the first is passed on, but sebsequently "chained" frames are put back into the read queue. Thus it was possible for a frame to go back into the jitter buffer where this would cause problems. This patch adds a flag to frames that are inserted into the channel's read queue after translation. The abstract jitter buffer code then checks for this flag and ignores any frames marked as such. Change-Id: I276c44edc9dcff61e606242f71274265c7779587 2016-11-06 06:30 +0000 [8cc1cd5df7] Sebastian Gutierrez * app_queue: Add QueueUpdate application. Add an application that allows tracking outbound calls using app_queue. ASTERISK-19862 Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e 2017-01-13 21:23 +0000 [f4e77a5678] Richard Mudgett * taskprocessor.c: Change when high water warning logged. The task processor queue reached X scheduled tasks message was originally intended to get logged only once per task processor to prevent spamming the log. This is no longer necessary since high and low water thresholds can better control when the message is logged. It is beneficial to generate the warning each time a task processor reaches the high water level because PJSIP stops processing new requests while any high water alert is active. Without this change you would have to enable at least debug level 3 logging to know about a repeated alert trigger. * Made generate the warning message whenever a task is pushed into the task processor that triggers the high water alert. * Appended 'again' to the warning for a repeated high water alert trigger. Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999 2017-01-10 05:54 +0000 [e0e502d9d2] Aaron An * res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt) Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter) always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and "AST_RTP_STAT_STRCPY". It should compare "combined" with "stat" not "current_stat". ASTERISK-26710 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15 2017-01-10 18:10 +0000 [0d53c91fba] George Joseph * debug_utilities: Create the ast_coredumper utility This utility allows easy manipulation of asterisk coredumps. * Configurable search paths and patterns for existing coredumps * Can generate a consistent coredump from the running instance * Can dump the lock_infos table from a coredump * Dumps backtraces to separate files... - thread apply 1 bt full -> .thread1.txt - thread apply all bt -> .brief.txt - thread apply all bt full -> .full.txt - lock_infos table -> .locks.txt * Can tarball corefiles and optionally delete them after processing * Can tarball results files and optionally delete them after processing * Converts ':' in coredump and results file names '-' to facilitate uploading. Jira for instance, won't accept file names with colons in them. Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. [1] For *BSDs, the "devel/gdb" package might have to be installed to get a recent gdb. The utility will check all instances of gdb it finds in $PATH and if one isn't found that can run python, it prints a friendly error. Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd (cherry picked from commit cb47b4556053cd50d9102eef913671ad0306062d) 2017-01-08 10:29 +0000 [e54c8aec34] George Joseph * pjproject_bundled: Fix compilation with MALLOC_DEBUG When MALLOC_DEBUG was specified, make was failing. Immediately remaking would work. The issues was in the ordering of the make dependencies. Change-Id: If6030b54fc693f3179f32bfd20c6b5d5f1b3f7cd 2017-01-05 06:11 +0000 [a7d856cd96] Joshua Colp * res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. This change implements SRV support for the IP based endpoint identifier module. All possible addresses through SRV are looked up and added as matches. If no SRV records are available a fallback to normal host resolution is done. If an IP address is provided then no SRV lookup occurs. This is configured using the "srv_lookups" option on the identify section and defaults to "yes". ASTERISK-26693 Change-Id: I6b641e275bf96629320efa8b479737062aed82ac 2016-11-06 06:37 +0000 [740ca862e4] Sebastian Gutierrez * app_queue: add new Service Level calculation Adds a new formula for SL2 and documentation ASTERISK-26559 Change-Id: I0970c620460507cd9d45b0d43600779c8915e770 2016-12-19 15:03 +0000 [d96e350256] Jonathan R. Rose * core/pbx: dialplan show - display filename/line# Adds the ability for extensions to be registered to include filename and line number so that dialplan show output can show the filename and line number of a config file responsible for generating a given extension. This only affects config modules that are written to use the new extension registering functions. In this patch, that only includes pbx_config, so extensions registered in extensions.conf and any included extension will be shown in this manner. Extensions registered in this manner will show the filename and line number *instead* of the registrar. ASTERISK-26658 #close Reported by: Jonathan R. Rose Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30 2016-12-22 09:13 +0000 [aea2285865] Alexander Traud * res_pjsip_session: Access SIPDOMAIN via Dialplan. This feature was available in the SIP channel driver chan_sip. For example, Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect and dial remote SIP-URIs. This change here sets the SIP destination domain of an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well. ASTERISK-26670 #close Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243 2017-01-04 05:50 +0000 [e220c11bec] Alexander Traud * chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND. After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats but remember the joint format. Cached formats contain default parameters, often create an empty fmtp line. However, a joint format might have passed format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and contain the resulting format parameters from a SDP negotiation. ASTERISK-26691 #close Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc 2017-01-03 15:14 +0000 [ceb9dae566] George Joseph * pjproject_bundled: Compile pjsua with max log level = 2 A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6. This allowed us to control the log level better from inside Asterisk. An unfortunate side effect of this was that the pjsua binary and python bindings were also compiled with log level set to 6 so whenever a testsuite test that uses pjsua runs, it spits out 6795 lines of debug in an instant even before the test starts. I believe this overruns the Jenkins capture buffer and prevents the test from properly terminating. In turn, this results in the testsuite just hanging until the job is killed. It's more frequent on the higher end agents because they can spit out the messages faster. Unfortunately, the messages are all spit out before we have control of the python pj.Lib instance where we can set logging levels so the only alternative was to actually compile pjsua and _pjsua.so with an overridden PJ_LOG_MAX_LEVEL. Although defining a lower max level was done in the Makefile, the define in config_site.h had to be wrapped with "#ifndef" so the change would take effect. Change-Id: I2af9e7d48dde1927279c586c9c725d868fe6f3ff 2016-12-22 16:00 +0000 [ae57652983] Joshua Colp * chan_pjsip: Use session for retrieving CHANNEL() information. The CHANNEL() dialplan function implementation for PJSIP allows querying of PJSIP specific information. This used the channel passed in to get the PJSIP session and associated information. It is possible for this channel to be masqueraded and end up as a different channel type by the time the information request is actually acted upon. This change retrieves the PJSIP session safely and accesses data from it (including channel). This provides a guarantee that the session and channel will not be altered when the request is being acted upon. ASTERISK-26673 Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6 2016-12-31 19:56 +0000 [386e3a01b3] Joshua Elson * res_pjsip: Fix known compact header issues ASTERISK-26684 #close Change-Id: Ifd7e401c45015119dd5e8421dbfe3afa6381744a 2016-12-30 06:59 +0000 [aad29b9bca] Martin Tomec * res_calendar: delete old calendars after reload When "fetch_again_at_reload" is set in config, we create now new object and thread for each reloaded calendar (with new configuration). Old calendar should be then unlinked, so the old thread can exit and free memory. ASTERISK-26683 Change-Id: Ic17fba9371c5a8b26a6bc54ea4957c13a32a343e 2016-12-30 09:10 +0000 [5a5953f98c] George Joseph * res_pjsip_refer: Handle compact Refer-To header. refer_incoming_refer_request needed to look for the "r" header as well as the "Refer-To" header. ASTERISK-26655 #close patches: refer_compact_fix.diff submitted by JoshE (license 6075) Change-Id: I610410a99b02427ea5db887aeb454d5f12c2259f 2016-12-23 12:11 +0000 [ac04e63ac2] Richard Mudgett * bridge_native_rtp.c: Minor code cleanups. In native_rtp_bridge_compatible_check() * Made one variable declaration per line. * Extracted if test assignment to make the test easier to see. * Made long if tests easier to see the combinatorial logic. * Added bridge id to a couple debug messages. Change-Id: I65bc5732aa7c9a2537f062f106fbea711cf2daad 2016-12-23 12:10 +0000 [da6f40c9ff] Richard Mudgett * bridge_native_rtp.c: Fix native rtp bridge data race. native_rtp_bridge_compatible() didn't lock the bridge channels before checking the channels for native bridging ability. As a result, one of the channel's native format capabilities structure got replaced out from under the native bridge check. Use of a stale pointer to freed memory causes bad things to happen. MALLOC_DEBUG, DO_CRASH, and the tests/channels/pjsip/transfers/blind_transfer/caller_direct_media testsuite test caught this. * Add missing channel locking in native_rtp_bridge_compatible(). Change-Id: If25fdb3ac8e85563c4857fb8216b3d9dc3d0fa53 2016-12-21 16:28 +0000 [b576b58d74] Richard Mudgett * res_rtp_asterisk.c: Fix uninitialized memory crash. ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor(). * Optimized out the 'us' struct variable. ASTERISK-26672 #close Change-Id: I4acea5dcdf0813da2c7d3e11c2d6067d160d17dc 2016-12-21 16:25 +0000 [67cc8499a2] Richard Mudgett * acl.c: Improve ast_ouraddrfor() diagnostic messages. * Made not generate strings unless they will actually be used. ASTERISK-26672 Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 2016-12-21 17:54 +0000 [67b47191e9] Richard Mudgett * chan_rtp.c: Fix uninitialized memory crash. unicast_rtp_request() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort the UnicastRTP channel request if it fails. ASTERISK-26672 Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0 2016-12-21 17:55 +0000 [2fc65173e5] Richard Mudgett * res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip(). We access uninitialized memory when the 'ourip' parameter does not have an initial guess to our IP address. ASTERISK-26672 Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15 2016-12-07 15:23 +0000 [8b7d252987] Richard Mudgett * res_rtp_asterisk.c: Fix off nominal memory leak. Change-Id: I95b1088d11244a2edae6607c12fbf33b38658a75 2016-12-14 02:21 +0000 [bab253ac9f] Tzafrir Cohen * Fixes to various issues reported by pyflakes Pyflake is a python (2) source checker. This patch fixes various (mostly trivial) errors and warnings it reports. Change-Id: Ia35c5ac61751b927814cf693994c632c412386ea 2016-12-09 12:23 +0000 [f461f65dea] Martin Tomec * app_queue: Ensure member is removed from pending when hanging up. In some cases member is added to pending_members, and the channel is hung up before any extension state change. So the member would stay in pending_members forever. So when we call do_hang, we should also remove member from pending. ASTERISK-26621 #close Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54 2016-12-18 15:23 +0000 [d29eb3b99d] George Joseph * pjproject_bundled: Make build single threaded There were just too many issues in various environments with multi threaded building of pjproject. It doesn't really speed things up anyway since asterisk is already being compiled in parallel. Change-Id: Ie5648fb91bb89b4224b6bf43a0daa1af793c4ce1 2016-12-08 20:00 +0000 [8fbb384ea2] Corey Farrell * chan_sip: Reorder unload_module to deal with stuck TCP threads. In some situations TCP threads may become frozen. This creates the possibility that Asterisk could segfault if they become unfrozen after chan_sip has been dlclose'd. This reorders the unload_module process to allow abort if threads do not exit within 5 seconds. High level order as follows: 1) Unregister from the core to stop new requests. 2) Signal threads to stop 3) Clear config based tables (but do not free the table itself). 4) Verify that threads have shutdown, cancel unload if not. 5) Clean all remaining resources. ASTERISK-26586 Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882 2016-12-16 01:32 +0000 [147b8e636e] David M. Lee * configure: fix with-pjproject-bundled The AC_ARG_WITH macro's shell variable is withval; not enableval. Purely coincidentally, the option would work when --enable-dev-mode is given. Also fixed a portability problem with bootstrap.sh, since -printf is not a portable option for find. Change-Id: I0f0e5b1a934b5af5737713834361e9c95b96b376 2016-12-15 13:25 +0000 [d27dee3cca] Richard Mudgett * autosupport: Add 'pjproject show buildopts' Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7 2016-12-14 14:21 +0000 [9404efa6f4] Richard Mudgett * chan_dahdi.c: Fix bounds check regression. Caused by ASTERISK-25494 Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb 2016-12-13 14:34 +0000 [45a5e2abc6] Richard Mudgett * res_pjsip: Add/update ERROR msg if invalid URI. ASTERISK-24499 Change-Id: Ie305153e47e922233b2ff24715e0e326e5fa3a6c 2016-12-12 18:38 +0000 [44e72c9d44] Richard Mudgett * MESSAGE: Flush Message/ast_msg_queue channel alert pipe. ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 2016-12-13 14:06 +0000 [19328de2ab] George Joseph * res_sorcery_memory_cache: Change an error to a debug message When a sorcery user calls ast_sorcery_delete on an object that may have already expired from the cache, res_sorcery_memory_cache spits out an ERROR. Since this can happen frequently and validly when an inbound registration expires after the cache entry expired, the errors are unnecessary and misleading. Changed to a debug/1. Change-Id: Idf3a67038c16e3da814cf612ff4d6d18ad29ecd7 2016-12-09 08:14 +0000 [31268e0a28] George Joseph * pjproject_bundled: Retry download if previously saved tarball is bad If a tarball is corrupted during download, the makefile will attempt to download it again. If the tarball somehow gets corrupted after it's downloaded however, the makefile was just failing. We now retry the download. ASTERISK-26653 #close Change-Id: I1b24d454852d80186f60c5a65dc4624ea8a1c359 2016-12-08 12:43 +0000 [4c6ba1dbba] Badalyan Vyacheslav * Fix typo in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb 2016-12-08 12:30 +0000 [934aa2c768] Badalyan Vyacheslav * res_pjsip: Fix 'A = B != C' kind. Consider reviewing the expression of the 'A = B != C' kind. The expression is calculated as following: 'A = (B != C)' Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d 2016-12-08 12:54 +0000 [51118e7d70] Badalyan Vyacheslav * chan_sip: Delete unneeded check P is always true. We check it before Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb 2016-12-08 12:58 +0000 [fe5be81821] Badalyan Vyacheslav * Small code cleanup in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a 2016-12-08 12:34 +0000 [149d8db96c] Badalyan Vyacheslav * Fix IO conversion bug Expression 'rlen < 0' is always false. Unsigned type value is never < 0. Change-Id: Id9f393ff25b009a6c4a6e40b95f561a9369e4585 2016-11-30 09:31 +0000 [c796f00c35] Walter Doekes * chan_sip: Do not allow non-SP/HTAB between header key and colon. RFC says SIP headers look like: HCOLON = *( SP / HTAB ) ":" SWS SWS = [LWS] ; sep whitespace LWS = [*WSP CRLF] 1*WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234 chan_sip implemented this: HCOLON = *( LOWCTL / SP ) ":" SWS LOWCTL = %x00-1F ; CTL without DEL This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with \x00-\x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To\x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change fixes so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. ASTERISK-26433 #close AST-2016-009 Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b 2016-11-14 18:18 +0000 [5c89604a32] Joshua Colp * res_format_attr_opus: Fix crash when fmtp contains spaces. When an opus offer or answer was received that contained an fmtp line with spaces between the attributes the module would fail to properly parse it and crash due to recursion. This change makes the module handle the space properly and also removes the recursion requirement. ASTERISK-26579 Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3 2016-12-06 14:54 +0000 [79b09b5f18] George Joseph * res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a 2016-12-07 14:22 +0000 [3b6e6cd01c] snuffy * tests_dns: Make DNS tests older nameser.h compatible Fix the tests for DNS to use older style nameser.h as in ASTERISK-26608. Tested on: OpenBSD 6.0, Debian 8 ASTERISK-26647 #close Change-Id: I285913c44202537c04b3ed09c015efa6e5f9052d 2016-12-06 16:45 +0000 [76d52dc228] Richard Mudgett * Bundled pjproject: Fix finding SIP transactions. Occasionally SIP message transactions are not found when they should be. In the particular case an incoming INVITE transaction is CANCELed but the INVITE transaction cannot be found so a 481 response is returned for the CANCEL. The problematic calls have a '_' character in the Via branch parameter. The problem is in the pjproject PJ_HASH_USE_OWN_TOLOWER feature's code. The problem with the "own tolower" code is that it does not calculate the same hash value as when the pj_tolower() function is used. The "own tolower" code will erroneously modify the ASCII characters '@', '[', '\\', ']', '^', and '_'. Calls to pj_hash_calc_tolower() can use the PJ_HASH_USE_OWN_TOLOWER substitute algorithm when enabled. Calls to pj_hash_get_lower(), pj_hash_set_lower(), and pj_hash_set_np_lower() call find_entry() which never uses the PJ_HASH_USE_OWN_TOLOWER algorithm. As a result you may not be able to find a hash tabled entry because the calculated hash values would differ. * Simply disable PJ_HASH_USE_OWN_TOLOWER. ASTERISK-26490 #close Change-Id: If89bfdb5f301b8b685881a9a2a6e0c3c5af32253 2016-12-01 16:49 +0000 [503006123a] Mark Michelson * http: Send headers and body in one write. This is a semi-regression caused by the iostreams change. Prior to iostreams, HTTP headers were written to a FILE handle using fprintf. Then the body was written using a call to fwrite(). Because of internal buffering, the result was that the HTTP headers and body would be sent out in a single write to the socket. With the change to iostreams, the HTTP headers are written using ast_iostream_printf(), which under the hood calls write(). The HTTP body calls ast_iostream_write(), which also calls write() under the hood. This results in two separate writes to the socket. Most HTTP client libraries out there will handle this change just fine. However, a few of our testsuite tests started failing because of the change. As a result, in order to reduce frustration for users, this change alters the HTTP code to write the headers and body in a single write operation. ASTERISK-26629 #close Reported by Joshua Colp Change-Id: Idc2d2fb3d9b3db14b8631a1e302244fa18b0e518 2016-12-06 10:56 +0000 [bf6423a336] Mark Michelson * Iostreams: Correct off-by-one error. ast_iostream_printf() attempts first to use a fixed-size buffer to perform its printf-like operation. If the fixed-size buffer is too small, then a heap allocation is used instead. The heap allocation in this case was exactly the length of the string to print. The issue here is that the ensuing call to vsnprintf() will print a NULL byte in the final space of the string. This meant that the final character was being chopped off the string and replaced with a NULL byte. For HTTP in particular, this caused problems because HTTP publishes the expected Contact-Length. This meant HTTP was publishing a length one character larger than what was actually present in the message. This patch corrects the issue by adding one to the allocation length. ASTERISK-26629 Reported by Joshua Colp Change-Id: Ib3c5f41e96833d0415cf000656ac368168add639 2016-12-06 12:06 +0000 [fe9f070885] George Joseph * pjproject_bundled: Fix missing inclusion of symbols Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to the CFLAGS. Not sure how they went missing. Also fixed an uninstall problem where we weren't removing the symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was there, I fixed it for libasteriskssl as well. Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556 2016-11-30 18:25 +0000 [4b3d3fc741] Richard Mudgett * res_pjsip_outbound_registration.c: Filter redundant statsd reporting. Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646 2016-06-28 16:26 +0000 [26c8552fff] Tzafrir Cohen * OpenSSL 1.1.0 support OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b 2016-11-22 11:20 +0000 [75230f4c01] Guido Falsi * res_rtp: Fix regression when IPv6 is not available. The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e 2016-11-23 18:27 +0000 [1dfa11b65c] Richard Mudgett * PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389 2016-11-30 10:48 +0000 [621d886ca7] Mark Michelson * Frame deferral: Re-queue deferred frames one-at-a-time. The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d 2016-11-15 15:01 +0000 [e5e887be53] Alexei Gradinari * chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d 2016-11-21 15:43 +0000 [ddc951060a] David Kerr * app_originate: Add option to execute gosub prior to dial Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57 2016-11-28 19:43 +0000 [0e214c4932] Eduardo S. Libardi * res_calendar_caldav: Add support reading gmail calendar The response from gmail calendar includes the string name "caldav:calendar-data". res_calendar_caldav implements the example included in RFC 4791: string "C:calendar-data". When reading the calendar, res_calendar_caldav compare the string and if does not match just discards the event. This commit compares the response to both strings, successfully loading gmail calendar events. Writing to gmail calendar is working prior to this fix. ASTERISK-26624 Reported by: Eduardo S. Libardi Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a 2016-11-28 15:12 +0000 [a3f48be0da] Matt Jordan * res/res_pjsip: Fix documentation whitespace issues Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0 2016-11-22 10:27 +0000 [0e15760795] Matt Jordan * res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise 'ws' when WebSockets are to be used as the transport. This applies to both secure and insecure WebSockets. There were two bugs in Asterisk with respect to this: (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for insecure websockets and 'wss' for secure websockets. While this would seem to make sense - since 'WS' and 'WSS' are used for the Via Transport parameter - this is not the case for the SIP URI. This patch corrects that by registering the secure websockets with pjproject using the shorthand 'WS', and by returning 'ws' when asked for the transport parameter. Note that in pjproject, it is perfectly valid to have multiple transports use the same shorthand. (2) In chan_sip, we return an upper-case version of the transport 'WS' instead of 'ws'. Since we should be strict in what we send and liberal in what we accept (within reason), this patch lower-cases the transport before appending it to the parameter. ASTERISK-24330 #close Reported by: cervajs, Inaki Baz Castillo Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42 2016-11-28 11:03 +0000 [8a68289766] George Joseph * build_tools: Fix download_externals to handle certified branches download_externals wasn't handling the "certified/13.x" version correctly. Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a 2016-11-28 07:36 +0000 [e3dae763ee] Joshua Colp * iostream: Move include of asterisk.h The asterisk.h header file needs to be included first or else some things go awry, such as: implicit declaration of function 'vasprintf' Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c 2016-11-26 10:57 +0000 [0b588778c0] Michael Kuron * chan_sip: Fix segfault during module unload If a TCP/TLS connection was pending (not accepted and not timed out) during unload of chan_sip, Asterisk would segfault when trying to send a signal to a thread whose thread ID hadn't been recorded yet. This commit fixes that by recording the thread ID before calling the blocking connect() syscall. This was a regression introduced by 776a14386a55b5425c7e9617eff8af8b45427144. The above wasn't enough to fix the segfault, which was now delayed to the point where connect() timed out. Therefore, it was necessary to also remove the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be used to interruput the connect() syscall. This was a regression introduced by 5d313f51b982a18f7321adcf7c7a4e822d8b2714. ASTERISK-26586 #close Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b 2016-11-23 14:52 +0000 [ead773f801] Dennis Guse * pbx_lua: On configuration errors report module load failure instead of decline. Switched from AST_MODULE_LOAD_DECLINE to AST_MODULE_LOAD_FAILURE. Therefore, if pbx_lua fails to load and pbx_lua is marked as required, Asterisk exits as expected. If extensions.lua cannot be opened, AST_MODULE_LOAD_DECLINE is reported. Change-Id: I8e5a0037e69b41743db60c568541ebb2f52a7a8f 2016-11-11 08:16 +0000 [d9b24cce0a] gestoip2 * res_rtp_asterisk: RTT miscalculation in RTCP When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f 2016-11-15 13:44 +0000 [635b0a0a55] Michael Kuron * tcptls: Use new certificate upon sip reload Previously, a TLS server socket would only be restarted upon sip reload if the bind address had changed. This commit adds checking for changes to TLS parameters like certificate, ciphers, etc. so they get picked up without requiring a reload of the entire chan_sip module. This does not affect open connections in any way, but new connections will use the new TLS parameters. The changes also apply to HTTP and Manager. ASTERISK-26604 #close Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6 2016-11-21 09:49 +0000 [abae3dc36e] George Joseph * pjproject_bundled: Use $(LIB_RT) for link of libasteriskpj libasteriskpj was hard coded to use -lrt but librt is linux specific so we now use the LIB_RT variable which gets set by configure. Change-Id: I41148884517e3031f7675a413d524c86e8614694 2016-11-19 16:19 +0000 [b546497fe0] snuffy * Add support for older name resolving version libraries like openBSD Fix support of OS's like openBSD that use an older nameser.h, this change reverts the defines to the older style which on other systems is found in nameser_compat.h Tested on openBSD 6.0, Debian 8 ASTERISK-26608 #close Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a 2016-11-18 09:46 +0000 [7a8d6bc81b] Mark Michelson * Bump ARI version to 2.0.0 In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665 2016-11-16 12:05 +0000 [d3f070c7a2] George Joseph * pjproject_bundled: Improve reliability of pjproject download The download process now has a timeout which will cause wget to retry if it stops retrieving data for 5 seconds and fetch and curl to timeout if the whole retrieval take smore than 30 seconds. If the tarball retrieval works, the MD5SUM file is retrieved from the downloads site and the md5 checksum is verified. If either the tarball retrieval or MD5SUM retrieval fails, or the checksums don't match, the entire process is retried once. If it fails again, any incomplete tarball is deleted. .DELETE_ON_ERROR: was also added to the Makefile. Not only does this delete the tarball on failure, it till also delete corrupted library files from the pjproject source directory should they fail to build correctly. Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and Ubuntu 14. Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1 2016-11-11 07:13 +0000 [e822a50f86] Mikheili Dautashvili * main/app.c: Transmit Silence on ControlPlayback pause ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8 2016-11-17 10:52 +0000 [d670ea6297] Mark Michelson * manager: update minor version Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: Idf84507354170400813cda780906c94c9f1b60b4 2016-11-17 08:25 +0000 [349e08cb48] Timo Teräs * codec_dahdi: Fix poll.h include. POSIX defines poll.h. sys/poll.h should not be used as it is c-library internal header which may or may not exist. Notably in musl including sys/poll.h generates warning of being incorrect. Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252 2016-11-16 20:24 +0000 [935f5d003b] George Joseph * build: Various OpenBSD issues OpenBSD's 'find' doesn't take the -delete argument so you have to pipe through 'xargs rm -rf'. 'echo -e' doesn't like \t starting a line. It just prints 't' which causes the libasteriskpj.exports file to be garbage. They were just cosmetic so they were removed. librt doesn't exist so the link of libasteriskpj.so fails. It's not actually needed for linux anyway so -lrt was removed from the link. res_rtp_asterisk was failing to load because of an undefined DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if so DTLSv1_method is used instead. ASTERISK-26608 Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c 2016-11-16 15:42 +0000 [dc8f99ee27] Mark Michelson * res_format_attr_opus: Fix fmtp generation. res_format_attr_opus assumed that the string being passed into it was empty. It tried to determine if the only thing it had written was a=fmtp: And if it had, it would reset the string. Its calculation was off when working with chan_sip, though. chan_sip passes the entire built SDP rather than an empty string. This resulted in always putting an empty fmtp line in the SDP. ASTERISK-26520 #close Reported by scgm11 Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5 2016-11-15 16:23 +0000 [ed9ced0531] Richard Mudgett * codec_opus: Fix warning when Opus negotiated but codec_opus not loaded. When Opus is negotiated but not loaded, the log is spammed with messages because the system does not know how to calculate the number of samples in a frame. * Suppress the warning by supplying a function that assumes 20ms of samples in the frame. For pass through support it doesn't really seem to matter what number of samples is returned anyway. ASTERISK-26605 #close Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f 2016-11-14 14:36 +0000 [0cd0e70c16] Richard Mudgett * res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak. Responding to authentication challenges leaks PJSIP memory pools. The leak was introduced with a pjproject 2.5.5 API change. https://trac.pjsip.org/repos/ticket/1929 changed the API usage of pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to clean up cached authentication allocations that get allocated with pjsip_auth_clt_reinit_req(). ASTERISK-26516 #close Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8 2016-11-15 12:01 +0000 [3017f09f22] George Joseph * file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type One of the code paths in __ast_file_read_dirs will only get executed if the OS doesn't support dirent->d_type OR if the filesystem the particular file is on doesn't support it. So, while standard Linux systems support the field, some filesystems like XFS do not. In this case, we need to call stat() to determine whether the directory entry is a file or directory so we append the filename to the supplied directory path and call stat. We forgot to truncate path back to just the directory afterwards though so we were passing a complete file name to the callback in the dir_name parameter instead of just the directory name. The logic has been re-written to only create a full_path if we need to call stat() or if we need to descend into another directory. Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba 2016-06-02 14:10 +0000 [070a51bf7c] Timo Teräs * Implement internal abstraction for iostreams fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85 2016-11-15 08:07 +0000 [d3b61a98f4] Joshua Colp * manager: Bump AMI version number. During the development of Asterisk 14 the behavior of the Command AMI action was altered such that the result was returned on lines with a prefix of "Output: ". While this was documented in the UPGRADE.txt file it is also reasonable that this should bump the AMI version number. ASTERISK-26556 Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42 2016-11-14 15:57 +0000 [edd7ae85e8] Matt Jordan * pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how many pairs of local/remote candidates will be made. If for some reason we reach this upper bound, ICE will generally fail and no media will flow between the browser and Asterisk. This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of pairs of candidates we'd theoretically allow, which is PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame Docker), this is far too low to allow WebRTC calls to succeed. Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed even when the system Asterisk was running on had quite a few virtual interfaces. Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55 2016-11-14 15:32 +0000 [cc86329228] Matt Jordan * apps/app_echo: Only relay a single video source change frame In 9785e8d0, app_echo was updated to relay video source updates to the channel for the purposes of displaying video in WebRTC tests. Unfortunately, this can cause a Kafkaesque nightmare if two or more Local channels are in a bridge together where their ends are in app_echo. When this situation occurs, a video update sent into app_echo will cause the video update to be relayed to the other Local channels, causing another round of video updates, etc. In not much time at all, the channel length queues will be overwhelmed, channel alert pipes will fail, and all hell will break loose as Asterisk merrily continues to throw more video update requests onto the channels. This patch updates app_echo to *only* relay a single video update. Once a video update has been made, all further video updates are dropped. This meets the intended purpose of the original patch: if we get a video update and we're in app_echo, go ahead and ask the sender to update themselves. However, once we've got that video stream sync'd up, don't keep spamming the world. Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74 2016-11-08 10:11 +0000 [a72ef38113] Matt Jordan * res/ari/resource_bridges: Add the ability to manipulate the video source In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621 2016-11-14 14:03 +0000 [7263a17ca0] George Joseph * channel: Fix issues in hangup scenarios caused by frame deferral ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 (cherry picked from commit 0be46aaf6b8b9eb5b0160ec591cdc2c6e1802a6d) 2016-11-14 13:55 +0000 [0dc4567133] George Joseph * Revert "Revert "channel: Use frame deferral API for safe sleep."" This reverts commit e5365dada5052b87275c048f6e29ac7d5e2b2415. Change-Id: Icc40cf0c7687454760762912dd29e4ae79e8e9ee 2016-11-14 13:55 +0000 [6d61f7bfd1] George Joseph * Revert "Revert "autoservice: Use frame deferral API"" This reverts commit edca6911f392f47c1a5a25d1d3a357c72b04a78a. Change-Id: I76030b87333a2c390cd05392b74b75678d78ddfa 2016-11-14 13:55 +0000 [f62c9c42fa] George Joseph * Revert "Revert "AGI: Only defer frames when in an interception routine."" This reverts commit 6bce938c2fcb60b7a77a0e997a6518860c0bfa39. Change-Id: Iadbf462bf2a52e8b2fa9ebc75b37b1f688ba51d9 2016-11-14 13:54 +0000 [2966fa5ad7] George Joseph * Revert "Revert "Add API for channel frame deferral."" This reverts commit fa749866c17f91860d3e9f89742eab3e6f03ecbc. Change-Id: Idcd1b88fa0766b1326dcc87d8905dbc314c71bd7 2016-11-11 10:45 +0000 [c6d755de11] Sebastien Duthil * res_ari: Add support for channel variables in ARI events. This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229 2016-11-14 12:16 +0000 [72da2ef9ff] George Joseph * cli: Fix ast_el_read_char to work with libedit >= 3.1 Libedit 3.1 is not build with unicode on as a default and so the prototype for the el_gets callback changed from expecting a char buffer to accepting a wchar buffer. If ast_el_read_char isn't changed, the cli reads garbage from teh terminal. Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and updated ast_el_read_char to use the HAVE_ define to detemrine whether to use char or wchar. ASTERISK-26592 #close Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a 2016-11-12 12:15 +0000 [97a75e3829] Tzafrir Cohen * Add support for building RADIUS with radcli Radcli is yet another RADIUS client library, generally compatible with freeradius and radiusclient-ng. This commit adds autoconf option for detecting it as well and changes cdr_radius and cel_radius to use its header file in that case. ASTERISK-26540 #close Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f 2016-11-10 10:57 +0000 [1bd49040c4] Joshua Colp * res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. When optimistic SRTP was on it was possible for us to still set up a call without an audio stream if an offer was received with required SRTP. This change makes it so this scenario will now fail with a 488 response. ASTERISK-26575 Change-Id: I7d14187037681f48879bd20319ac79d0877318f3 2016-11-11 02:41 +0000 [dfb951817f] Igor Goncharovskiy * Fix closing rtp ports after call finished in chan_unistim. Fix ASTERISK-26565 by adding ast_rtp_instance_stop before rtp instance destroy for chan_unistim. Also several fixes for displayed text translation. Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc 2016-11-11 00:29 +0000 [939dcf66b0] Timo Teräs * addons/chan_mobile: do not use strerror_r The two reasons why it might be used are that some systems do not implement strerror in thread safe manner, and that strerror_r returns the error code in the string in case there's no error message. However, all of asterisk elsewhere uses strerror() and assumes it to be thread safe. And in chan_mobile the errno is also explicitly printed so neither of the above reasons are valid. The reasoning to remove usage is that there are actually two versions of strerror_r: XSI and GNU. They are incompatible in their return value, and there's no easy way to figure out which one is being used. glibc gives you the GNU version if _GNU_SOURCE is defined, but the same feature test macro is needed for other symbols. On all other systems you assumedly get XSI symbol, and compilation warnings as well as non-working error printing. Thus the easiest solution is to just remove strerror_r and use strerror as rest of the code. Alternative is to introduce ast_strerror in separate translation unit so it can request the XSI symbol in glibc case, and replace all usage of strerror. Change-Id: I84d35225b5642d85d48bc35fdf399afbae28a91d 2016-09-23 17:54 +0000 [338f35edcc] Richard Mudgett * res_pjsip.c: Rework endpt_send_request() req_wrapper code. * Don't hold the req_wrapper lock too long in endpt_send_request(). We could block the PJSIP monitor thread if the timeout timer expires. sip_get_tpselector_from_endpoint() does a sorcery access that could take awhile accessing a database. pjsip_endpt_send_request() might take awhile if selecting a transport. * Shorten the time that the req_wrapper lock is held in the callback functions. * Simplify endpt_send_request() req_wrapper->timeout code. * Removed some redundant req_wrapper->timeout_timer->id assignments. Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9 2016-09-21 15:10 +0000 [bb196323f9] Richard Mudgett * res_pjsip: Fix tdata leaks in off nominal paths. Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b 2016-10-24 12:41 +0000 [9df59d9ff4] Richard Mudgett * res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94 2016-11-10 13:38 +0000 [73524bde9c] C.J. Collier * chan_sip: Fix typo and re-wrap surrounding docs Correct typo of end-pints to end-points Re-wrap session timer parameter docs to max 80 chars wide; this eases reading on terminals with lower resolution, commonly the case for those with visual impairments. ASTERISK-26573 Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b Signed-off-by: C.J. Collier 2016-11-09 15:14 +0000 [bdb6d928c5] Joshua Colp * res_pjsip: Perform resolution when explicit IPv6 transport is used. This change fixes the SIP resolver such that if an IPv6 transport is explicitly used it will resolve NAPTR, SRV, and AAAA records. You can explicitly use one by specifying it on an endpoint. ASTERISK-26571 Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36 2016-11-10 08:33 +0000 [93a0de1f0e] Joshua Colp * app_queue: Add mention of 'ABANDON' variable to CHANGES. ASTERISK-26558 Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e 2016-11-10 07:34 +0000 [fa749866c1] George Joseph * Revert "Add API for channel frame deferral." This reverts commit f073f648b87d45e4729969fd2d83695c300757d1. Multiple testsuite failures were detected after the fact. Change-Id: I968c380418bf65c7166f6ecff30fe8e247ea6682 2016-11-10 07:33 +0000 [6bce938c2f] George Joseph * Revert "AGI: Only defer frames when in an interception routine." This reverts commit 28926d1c81540bbeb16802814d3f2e63c2347bd2. Multiple testsuite failures were detected after the fact. Change-Id: I8d4f5ccbb421a351d616254844ae7e5a31053edb 2016-11-10 07:32 +0000 [edca6911f3] George Joseph * Revert "autoservice: Use frame deferral API" This reverts commit afef1b8e4a311d33b3e485b9bab3c6e7fd13fbc9. Multiple testsuite failures were detected after the fact. Change-Id: Ib4cb0c0a6475681ce817f71b4050be25640ab67f 2016-11-10 07:31 +0000 [e5365dada5] George Joseph * Revert "channel: Use frame deferral API for safe sleep." This reverts commit 392202304d248147378f1e16f1f012285dc1221f. Multiple testsuite issues were discovered after the fact. Change-Id: I848c4196dca2994b1a368087004326ea354cff95 2016-11-09 18:18 +0000 [edea41126b] George Joseph * build: Fix default values for some SANITIZER options 2 of the sanitizers didn't have default values so in systems that don't support sanitizers menuselect would spit out warnings. They were harmless but confusing. They've now been set to "0". Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58 2016-11-06 06:04 +0000 [4e8ab6cda9] Sebastian Gutierrez * app_queue: new variable set when abandoned sets the variable ABANDONED to TRUE if the call was not answered. ASTERISK-26558 Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3 2016-11-08 10:48 +0000 [e5860ce07d] Mark Michelson * res_pjsip_session: Do not call session supplements when it's too late. res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 2016-11-03 16:46 +0000 [392202304d] Mark Michelson * channel: Use frame deferral API for safe sleep. This is another case where manual frame deferral can be replaced with centralized routines instead. Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e (cherry picked from commit d149c4b9e07eeb880d8428ad52c6fdb315cc15f5) 2016-11-03 16:46 +0000 [afef1b8e4a] Mark Michelson * autoservice: Use frame deferral API Rather than use manual frame deferral, just let the channel API do it for us. ASTERISK-26343 Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49 (cherry picked from commit 8ba3e2fc27f9966b8c7ce75c1eca6208613a9315) 2016-11-03 16:42 +0000 [28926d1c81] Mark Michelson * AGI: Only defer frames when in an interception routine. AGI recently was modified to defer important frames. This was because when AGI was used in a connected line interception routine, the resulting connected line frame would end up getting discarded by the AGI. However, this caused bad behavior in other cases. Specifically, during a transfer, if someone attempted to manually set the Caller ID on a channel in an AGI, the deferred connected line frame would end up overwriting what had been manually set in the AGI. Since the initial issue was specific to interception routines, this change removes the manual frame deferral from AGI and instead uses the new frame deferral API in interception routines. ASTERISK-26343 #close Reported by Morton Tryfoss Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208 2016-11-03 16:36 +0000 [f073f648b8] Mark Michelson * Add API for channel frame deferral. There are several places in Asterisk that have duplicated logic for deferring important frames until later. This commit adds a couple of API calls to facilitate this automatically. ast_channel_start_defer_frames(): Future reads of deferrable frames on this channel will be deferred until later. ast_channel_stop_defer_frames(): Any frames that have been deferred get requeued onto the channel. ASTERISK-26343 Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641 2016-11-02 10:52 +0000 [d30415bfa1] Joshua Colp * res_stasis: Don't unsubscribe from a NULL bridge. A NULL bridge has special meaning in res_stasis for unsubscribing. It means that a subscription to ALL bridges should be removed. This should not be done as part of the normal subscription management in the res_stasis channel loop. ASTERISK-26468 Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0 2016-11-03 07:42 +0000 [0a698cd932] Alexander Anikin * chan_ooh323: Fixes to work right with Cisco devices Changed output packets queue processing algo to one read-one write instead of all read-all send Remove h.245 tunneling parameter from ReleaseComplete packet ASTERISK-24400 #close Reported by: Dmitry Melekhov Tested by: Dmitry Melekhov Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6 2016-11-03 13:10 +0000 [a1cdc3891a] Alexander Anikin * chan_ooh323: reset rrq count on gk registration reset registration attempts count on success registration on gatekeeper Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336 2016-11-06 05:40 +0000 [b2b5f9d897] frahaase * ast_format: Adds an identifier for interleaved audio formats to the ast_format Adds an identifier (with a getter and setter) to detect channels with interleaved audio. This is needed by the binaural bridge_softmix patch (ASTERISK-26292) and was already discussed here: http://lists.digium.com/pipermail/asterisk-dev/2016-October/075900.html The identifier can be set during fmtp parsing (to be seen in the res_format_attr_opus.c change). ASTERISK-26292 Change-Id: I359801cc5f98c35671c48dabc81a7f4ee1183d63 2016-11-06 03:46 +0000 [fbbbd0add9] Michael Kuron * automon: restore mixing of the both channels after recording stops This is a regression over Asterisk 11, introduced by 2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via the automon DTMF code would automatically be mixed together using sox because app_monitor would be called with the m option. This commit restores this behavior. Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759 2016-11-04 15:42 +0000 [367d4903cc] Matt Jordan * res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems Not surprisingly, using Respoke (and possibly other systems) it is possible to blow past the 16k limit for a WebSocket packet size. This patch bumps it up to 32k, which, at least for Respoke, is sufficient. For now. Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that matter), this patch adds a LOW_MEMORY directive that sets the buffer to 8k for systems who have asked for their reduced memory availability to be considered. Change-Id: Id235902537091b58608196844dc4b045e383cd2e 2016-11-04 15:40 +0000 [7a449b6819] Matt Jordan * res_stasis: Set a video source mode on Stasis created bridges When a bridge is created via ARI (through res_stasis), no video source mode is set by default. As a result, any endpoint sending video media won't ever see any video reflected back to it. This patch defaults a bridge to a 'follow the talker' video mode. Further work can be done to add routes that allow for the video mode to be controlled through the /bridges resource. Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866 2016-11-04 15:37 +0000 [bbe943729a] Matt Jordan * main/bridge_channel: Fix channel reference leak on video source When a channel is made the video source, the bridge holds a reference to it. Whenever the video source changes, that reference is released. However, a ref leak does occur if the channel leaves the bridge (such as being hung up) while it is the video source, as the bridge never releases the ref in such a case. This patch adds a line to the bridge_channel_internal_join routine such that, when a channel finishes its time in the bridge, it notifies the bridge via ast_bridge_remove_video_src that if it is a video source its reference should be released. ASTERISK-26555 #close Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a 2016-11-04 15:36 +0000 [a70d6dba8c] Matt Jordan * main/bridge: Add some verbose logging for video source changes It's actually quite useful to see the source of a video stream change. This doesn't happen terribly often, even with talk detection - but when it does, it's nice to know which channel is now providing your video stream. As a verbose 5 level message, it shouldn't be terribly spammy or costly to have, and is 'lower level' then most other verbose messages that the bridge system emits. ASTERISK-26555 Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c 2016-11-04 15:33 +0000 [fb17b630a5] Matt Jordan * bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source WebRTC clients really, really want to know the SSRC of the media they're getting. Changing the SSRC is generally not a good thing. bridge_softmix, starting in Asterisk 12, started changing the SSRC of parties as they joined or left the bridge. With most phones, this isn't a problem: phones just play back the stream they're getting. With WebRTC clients, however, the SSRC is tied to a media stream that may be negotiated. When a new SSRC just shows up, the media can be dropped. As it turns out, the SSRC change shouldn't even be necessary. From the perspective of the client, it's still talking to Asterisk with the same media stream: why indicate that the far party has suddenly changed to a different source of media? This patch opts to just remove the SSRC changes. With this patch, video clients that join/leave a softmix bridge actually get the video stream instead of freaking out. ASTERISK-26555 Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf 2016-10-28 15:11 +0000 [70d5f90e3d] Kevin Harwell * stasis_recording/stored: remove calls to deprecated readdir_r function. The readdir_r function has been deprecated and should no longer be used. This patch removes the readdir_r dependency (replaced it with readdir) and also moves the directory search code to a more centralized spot (file.c) Also removed a strict dependency on the dirent structure's d_type field as it is not portable. The code now checks to see if the value is available. If so, it tries to use it, but defaults back to using the stats function if necessary. Lastly, for most implementations of readdir it *should* be thread-safe to make concurrent calls to it as long as different directory streams are specified. glibc falls into this category. However, since it is possible that there exist some implementations that are not safe, locking has been added for those other than glibc. ASTERISK-26412 ASTERISK-26509 #close Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba 2016-11-04 10:57 +0000 [bf01ff53f8] Kevin Harwell * Revert "chan_sip: Fix lastrtprx always updated" This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d 2016-11-03 13:45 +0000 [1504194215] Alexander Anikin * chan_ooh323: Fix infinite loop on read second part of H.225 packet Fix logic on read second part of H.225 packet. There was infinite loop on wrong connections due to read before poll. Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff 2016-11-03 11:55 +0000 [78dc6ceaf6] George Joseph * pjproject_bundled: Fix issue with libasteriskpj needing libresample libresample is only needed by pjproject if we're building pjsua, which we only do if TEST_FRAMEWORK is selected. It's required by pjsua to process audio which is needed by some testsuite tests. Unfortunately, pjproject relies on a newer version of libresample than the version that ships by most distros so we need to compile the version that's bundled with pjproject. Since we only need it for pjsua, we DON'T want it's symbols exposed when we actually build asterisk. There was a problem however... TEST_FRAMEWORK is only known AFTER we've already run ./configure on both asterisk and pjproject but pjproject's ./configure needs to test it to know whether to set up to build libresample or not. The previous way of figuring this out was to always tell ./configure "yes" but not actually build the library. This caused an issue where building libasteriskpj was being told to include libresample but it wasn't actually there. The solution is to still do a default pjproject configure during an asterisk ./configure but if makeopts or menuselect.makeopts changes subsequently, we now reconfigure pjproject, taking into account the current state of TEST_FRAMEWORK. Previously, if makeopts or menuselect.makeopts changed, only a recompile of pjproject was done. Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685 2016-11-01 19:48 +0000 [0904c1f4cc] Sebastian Gutierrez * chan_sip: add missing account code Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482 2016-11-02 09:15 +0000 [4de5454ef1] Joshua Colp * app_dial: Fix incorrect device state when channel is picked up. Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f 2016-09-13 04:08 +0000 [9ac53877f6] Alexander Traud * rtp_engine: Allow more than 32 dynamic payload types. Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, this change utilizes payload types in the range between 35 and 63 giving room for another 29 payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964 2016-11-02 05:05 +0000 [6a99f007d6] Tzafrir Cohen * autoconf: more variants for OSARCH linux-gnu There are quite a few odd GNU/Linux platforms. Just call all of them linux-gnu. Specifically this fixes building the Debian platforms mips64el and x32. And maybe also others. ASTERISK-26546 #close Change-Id: I06ec4bd7f0ee1c84b6b24d81538223b07c4174b1 2016-11-01 13:13 +0000 [f29b8d62bb] Richard Mudgett * bundled pjproject: Fix DNS write to freed memory. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patch below fixes a write to freed memory under cartain DNS lookup conditions. 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch ASTERISK-26516 Reported by: Richard Mudgett Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5 2016-11-01 06:56 +0000 [6233e146c6] Joshua Colp * res_pjsip_sdp_rtp: Limit number of formats to defined maximum. The res_pjsip_sdp_rtp module did not restrict the number of formats added to a media stream in the SDP to the defined limit. If allow=all was used with additional loaded codecs this could result in the next media stream being overwritten some. This change restricts the module to limit it to the defined maximum and also increases the maximum in our bundled pjproject. ASTERISK-26541 #close Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8 2016-10-31 17:35 +0000 [8060cd1ec1] Kevin Harwell * codecs.conf.sample: Add sample and option descriptions for codec_opus codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b 2016-10-20 07:27 +0000 [c30d677333] Matt Jordan * res/stasis: Add CLI commands for displaying/debugging ARI apps This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5 2016-11-01 08:32 +0000 [2526dff94d] Grachev Sergey * chan_sip: Incorrect display option Outbound reg. retry 403 If in sip.conf (general section) set option register_retry_403=no, the command "sip show settings" return value: Outbound reg. retry 403:0 If in sip.conf (general section) set option register_retry_403=yes, the command "sip show settings" return value: Outbound reg. retry 403:-1 * In static char "sip show settings" for "Outbound.reg. retry 403" option use AST_CLI_YESNO ASTERISK-26476 #close Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9 2016-11-01 04:18 +0000 [ed08811e64] Tzafrir Cohen * netsock.c: fix includes for HURD ASTERISK-25070 Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814 2016-11-01 04:00 +0000 [69fed26deb] Tzafrir Cohen * define PATH_MAX for HURD PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD define it to a constant. It is indeed not safe to assume there won't be longer paths and Asterisk generally does err safely on such cases. So even for HURD we'll just pretend PATH_MAX is 4096. ASTERISK-25070 #close Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3 2016-10-31 16:12 +0000 [f27f837a9f] George Joseph * pjproject_bundled: Fix compile of pjsua so it handles audio In order for pjsua and its python binding to actually negotiate audio for the testsuite tests, it needs g711 and resample. The pj* libraries themselves do not. Unfortunately, pjproject relies on a brand new libresample that most distros don't ship so we need to use the libresample already bundled with pjproject. Only the pjsua executable and the _pjsua.so python library are linked with it so it shouldn't interfere with asterisk itself. Also it was pointed out that apply_patches couldn't handle multiple patches that depended on each other during the dry-run, so the dry-run was removed. Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098 2016-10-31 13:46 +0000 [1648ca06c3] Etienne Lessard * manager: Add documentation for NewConnectedLine event. The NewConnectedLine event has been added by commit fe7671f, but the documentation was missing. ASTERISK-26537 #close Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6 2016-10-30 13:33 +0000 [273debd261] Corey Farrell * vector: Prevent NULL argument to memcpy. Headers declare that memcpy does not accept NULL argument for the first two parameters. Add a conditional block to prevent memcpy and ast_free from running on vectors with NULL element array. ASTERISK-26526 #close Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71 2016-10-29 10:19 +0000 [ad60927a40] Corey Farrell * astobj2: Declare private variable data_size for AO2_DEBUG only. Every ao2 object contains storage for a private variable data_size, though the value is never read if AO2_DEBUG is disabled. This change makes the variable conditional, reducing memory usage. ASTERISK-26524 #close Change-Id: If859929e507676ebc58b0f84247a4231e11da07f 2016-10-28 14:55 +0000 [6feee22e09] Richard Mudgett * bundled pjproject: Crashes while resolving DNS names. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patches below fix the DNS lookup race condition crash caused by attempting to send the same message twice for the single DNS lookup. 0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch 0006-r5473-svn-backport-Fix-pending-query.patch The patch below removes a cached DNS response from the hash table when another thread is referencing the old entry. The table still contained the entry when it was destroyed which can result in inexplicable crashes. 0006-r5475-svn-backport-Remove-DNS-cache-entry.patch ASTERISK-26344 #close Reported by: Ian Gilmour ASTERISK-26387 #close Reported by: Harley Peters Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4 2016-10-28 16:59 +0000 [12bdde6a6c] George Joseph * pjproject_bundled: Fix issue where "/version.mak" wasn't found main/Makefile includes third-party/pjproject/build.mak but doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak" evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile before the include. Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604 2016-10-28 13:30 +0000 [9d8b9b6ca5] Matt Krokosz * res_pjsip_outbound_publish: Fix crash when publishing device state. While publishing device state between multiple instances of Asterisk, a crash will sporadically occur under high CPS which looks to be a race condition operating on the publisher queue. ASTERISK-26506 Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6 2016-10-27 21:49 +0000 [d6ad867897] Corey Farrell * Fix shutdown crash caused by modules being left open. It is only safe to run ast_register_cleanup callbacks when all modules have been unloaded. Previously these callbacks were run during graceful shutdown, making it possible to crash during shutdown. ASTERISK-26513 #close Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21 2016-10-28 09:50 +0000 [badd38f031] Rusty Newton * SAC documentation: don't specify transports for endpoints and registrations Removing explicit transport definition for endpoints and registrations. It isn't necessary and isn't generally advised. ASTERISK-26514 #close Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb 2016-10-18 09:06 +0000 [0646b48ece] Tzafrir Cohen * chan_dahdi: remove by_name support Support for referring to DAHDI channels by logical names was added in (FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support of refering to channels by name. While technically usable, it has never been properly supported in dahdi-tools, as using it would require many changes at the Asterisk level. Instead logical mapping was added at the kernel level. Thus it seems that refering to DAHDI channels by name is not really used by anyone, and therefore should probably be removed. Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485 2016-10-26 18:48 +0000 [4f45d62653] George Joseph * pjproject_bundled: Remove usage of tar's --strip-components option Older versions of tar don't support the --strip-components option so instead of doing 'tar --strip-components=1 -C source', we now just untar to the tarball's root directory (pjproject-) and rename that directory to 'source'. Also fixed an issue where the pjproject source directory is a hard coded absolute pathname. ASTERISK-26510 #close ASTERISK-22480 #close Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0 2016-10-26 21:40 +0000 [a6e5bae3ef] Corey Farrell * Remove ASTERISK_REGISTER_FILE. ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 2016-10-27 08:07 +0000 [6993f3c9c3] Joshua Colp * res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. The res_pjsip_caller_id module wrongly assumed that a saved From header would always exist on sessions. This is true until an inbound call is received and a session timer causes an UPDATE to be sent. In this case there will be no saved From header and a crash will occur. This change makes it fall back to the From header of the outgoing request if no saved From header is present. ASTERISK-26307 #close Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa 2016-10-26 07:51 +0000 [95062fe220] Joshua Colp * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice mailbox was not cleared, causing it to try to free garbage. ASTERISK-26503 #close Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3 2016-10-23 07:38 +0000 [aed6c219a3] Joshua Colp * pjsip: Fix a few media bugs with reinvites and asymmetric payloads. When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc 2016-10-26 06:32 +0000 [7925f60cd9] Joshua Colp * res_pjsip_sdp_rtp: Fix address family of explicit media_address. When an explicit media_address is provided the address family in the SDP needs to be set to reflect it. ASTERISK-26309 Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79 2016-10-25 11:20 +0000 [802bbf8752] George Joseph * test_astobj2_thrash: Fix multithreaded issues The test uses 4 threads to grow, count, lookup and shrink 15K objects in a container. If there's only 1 execution engine available, the test will complete in <50ms. If each threads gets its own execution engine, the test may timeout after 60 seconds because the count thread does a locked ao2_callback on the whole container in a tight loop with only a sched_yield to give up time. The lock contention makes the test execution times wildly variable and mostly timeout. 2 execution engines are OK, 3 results in about 33% failure rate and >=4 causes a 80% failure rate. To fix, the sched_yield was changed to a usleep(500). Also, the number of buckets specified for the container was an even number so that was changed to the next prime number greater than (MAX_HASH_ENTRIES / 100). That's 151 currently. Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77 2016-10-18 09:04 +0000 [2b9ad3a5f7] Alexei Gradinari * chan_pjsip: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk. This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref to inform pjproject that an INVITE session is in use. ASTERISK-26482 #close Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33 2016-10-10 11:49 +0000 [01d1d3763f] Badalyan Vyacheslav * cdr_radius,cel_radius: Fix old memleak in unload - Call "rc_openlog" optional. If you do not call, you will simply NULL instead of a name. - On the one PID can be only one syslog channel. And it can already be run in logger.c - Calling rc_openlog we assigns a new name for the channel syslog. This unexpected behavior for logger.c. Most lesser evil, is to agree on a NULL name syslog if the channel was not launched in logger.c. It also solves the problem of memory leaks. ASTERISK-26455 #close Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd 2016-10-24 10:55 +0000 [16c23b57c7] George Joseph * pjproject_bundled: Fixed various build issues * CFLAGS is now properly set when using older gcc. * All third-party pjproject targets have been removed. This fixes an issue with older libsrtp in some distros. * Manually removing the source directory now causes a rebuild. * EXTERNALS_CACHE_DIR is now properly checked. * Whitespace fixes. Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60 2016-10-24 14:13 +0000 [1d277e7cb6] Pascal Cadotte Michaud * typo: s/paranthesis/parenthesis/ in a comment Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30 2016-09-19 06:13 +0000 [403c4f5833] Joshua Colp * pjsip: Support dual stack automatically. This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d 2016-10-19 12:05 +0000 [3bd76dd679] Mark Michelson * ARI: Add duplicate channel ID checking for channel creation. This is similar to what is done for origination, but for the 14 and up channel creation method. When attempting to create a channel, if a channel ID is specified and a channel already exists with that ID, then a 409 is returned. Change-Id: I77f9253278c6947939c418073b6b31065489187c 2016-10-17 14:18 +0000 [e459b8dadf] Mark Michelson * ARI: Detect duplicate channel IDs ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06 2016-10-19 17:53 +0000 [e03364c40a] snuffy * Fix issue with CLI not returning to prompt after running "features show" ASTERISK-26444 #close Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8 2016-10-04 18:24 +0000 [3e96d491d0] Michael Walton * res_rtp_asterisk: Add ice_blacklist option Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the form ice_blacklist = , e.g. ice_blacklist = 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay discovery. This is useful for optimizing the ICE process where a system has multiple host address ranges and/or physical interfaces and certain of them are not expected to be used for RTP. Multiple ice_blacklist configuration lines may be used. If left unconfigured, all discovered host addresses are used, as per previous behavior. Documention in rtp.conf.sample. ASTERISK-26418 #close Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9 2016-10-18 16:30 +0000 [f14ef51ead] Mark Michelson * CDR: Alter destruction pattern for CDR chains. CDRs form chains. When the root of the chain is destroyed, it then unreferences the next CDR in the chain. That CDR is destroyed, and it then unreferences the next CDR in the chain. This repeats until the end of the chain is reached. While this typically does not cause any sort of problems, it is possible in strange scenarios for the CDR chain to grow way longer than expected. In such a scenario, the destruction pattern can result in a stack overflow. This patch fixes the problem by switching from a recursive pattern to an iterative pattern for destruction. When the root CDR is destroyed, it is responsible for iterating over the rest of the CDRs and unreferencing each one. Other CDRs in the chain, since they are not the root, will simply destroy themselves and be done. This causes the stack depth not to increase. ASTERISK-26421 #close Reported by Andrew Nagy Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8 2016-10-18 11:51 +0000 [f31772ec20] Joshua Colp * ari: Update model validator based on addition of asterisk_id. ASTERISK-26470 Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541 2016-09-11 10:13 +0000 [18a6f250e2] Tzafrir Cohen * menuselect: invalid test for GTK2 configuire.ac was only checking for the existence of pkg-config and not the gtk2 package itself. Now it calls AST_PKG_CONFIG_CHECK for gtk+-2.0. ASTERISK-26356 #close Change-Id: I93e9d0166341f0e7f84b52955bb6f81da42f2ef6 2016-10-18 03:01 +0000 [a43ee21211] Alexander Traud * cli: Auto-complete File not Module for core set debug. Since Asterisk 1.8, the command "core set debug" on the command-line interface asks not for a file (.c) but a module name. This change shows modules (.so) on the auto-completion via a tabulator or the question mark. Now, when you partially type a module name, TAB or ?, you get the correct candidiates. ASTERISK-26480 Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0 2016-08-12 11:22 +0000 [dce31f90ba] frahaase * Binaural synthesis (confbridge): On/off setting for binaural synthesis. Adds setting to confbridge.conf (binaural_active) that determines if binaural synthesis can be available in bridge_softmix. ASTERISK-26292 Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db 2016-10-17 11:39 +0000 [2a808b2fa6] George Joseph * pjproject_bundled: Add patch to address SSL crash Addresses crashes when an attempt is made to operate on an SSL socket after the socket has been closed. ASTERISK-26477 #close Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 2016-10-13 14:09 +0000 [973e57d5ce] Leandro Dardini * app_queue: Added initialization for "context" parameter When using Asterisk Realtime Architecture, empty fields are skipped and the default values are used. If the "context" parameter in queue was set and then cleared from the database, the old value remains in memory and it continues to be used. This change initialize the "context" parameter with an empty value, allowing clearing the parameter. ASTERISK-26462 #close Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905 2016-10-15 20:05 +0000 [dd5129d84a] Matt Jordan * res/ari: Add the Asterisk EID field to outgoing events This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d 2016-10-13 02:06 +0000 [2b03017022] Moises Silva * chan_rtp: Set a sane default rtp engine for unicast. ASTERISK-26439 Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011 2016-10-16 17:25 +0000 [6651c66e68] George Joseph * utils.c: Fix ast_set_default_eid for multiple platforms ast_set_default_eid was searching for ethX, emX, enoX, ensX and even pciD#U interface names. While this was a good attempt, it wasn't inclusive enough to capture interfaces like enp6s0 or ens6d1, etc. Rather than relying on interface names, we now simply find the first interface returned by the OS that has a hardware address and that address isn't all 0x00 or all 0xff. The code IS different for BSD, Solaris and Linux based on what method is available for enumerating interfaces. Tested on: FreeBSD9 CentOS6 Ubuntu14 Fedora24 I was unable to test on Solaris at this time but the code for Solaris is used elsewhere at Digium. Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72 2016-10-15 04:58 +0000 [e9315791b3] Michael Kuron * chan_sip: Only send video on outgoing channel if incoming channel supports it Previously, the settings videosupport=always and videosupport=yes behaved identically and unconditionally caused a video offer to be sent in the SDP on an outgoing call. This was a regression introduced with commit 5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1. This commit restores correct behavior: videosupport=always causes a video offer to be sent unconditionally, while videosupport=yes will only offer video on an outbound channel if the incoming channel it is bridged to also supports video. That way, the device receiving the outgoing call can display the correct user interface elements for audio or video and will not unnecessarily show a blank video window on an audio-only call. ASTERISK-17470 #close Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae 2016-10-14 00:18 +0000 [aa39a87697] Corey Farrell * Fix issues with bundled pjproject cached download. Previously when testing I had a preexisting makeopts in ASTTOPDIR. The ordering of configure.ac causes --with-externals-cache to be processed after third-party configure. In cases where the Asterisk clone is cleaned it would cause pjproject to be downloaded to /tmp. This moves processing of the externals cache and sounds cache to happen before third-party configure. This also addresses a possible issue with the third-party Makefile. If TMPDIR is set by the environment it would override the path given to --with-externals-cache. ASTERISK-26416 Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d 2016-10-12 16:24 +0000 [9c49b96374] Richard Mudgett * Audit ast_json_pack() calls for needed UTF-8 checks. Added needed UTF-8 checks before constructing json objects in various files for strings obtained outside the system. In this case string values from a channel driver's peer and not from the user setting channel variables. * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json object construction. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096 2016-10-12 16:20 +0000 [774d5f7ef7] Richard Mudgett * json: Check party id name, number, subaddresses for UTF-8. * Updated unit test as ast_json_name_number() is now NULL tolerant. ASTERISK-26466 #close Reported by: Richard Mudgett Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6 2016-10-11 18:14 +0000 [1c4c6c082d] Richard Mudgett * json: Add UTF-8 check call. Since the json library does not make the check function public we recreate/copy the function in our interface module. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99 2016-10-12 17:42 +0000 [6fe5202c2c] Richard Mudgett * aoc.c: Whitespace cleanup * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null() when creating the type json object. The ref is a noop. Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a 2016-10-12 16:22 +0000 [c3bf1632cd] Richard Mudgett * app_minivm.c: Fix malformed ast_json_pack() call. Change-Id: I082b239022fac462666e52a14a44304748908dc0 2016-10-12 17:27 +0000 [9c54964dc5] Richard Mudgett * app_queue.c: Fix clearing of pause reason string. The pause reason is not always cleared when it should be cleared. * Made set_queue_member_pause() always clear pause reason if not pausing with a reason string. Change-Id: I993dad19626ec017478a230e980989438b778c53 2016-10-12 16:30 +0000 [3b3d06884c] George Joseph * res_config_mysql: Fix several issues related to recent table changes Unlike any of the other database drivers, res_config_mysql checks that the table definition matches the requirements for every insert and update statement. Since all requirements are forced to 'char', any column that isn't a char, like ps_contacts' expiration_time, qualify_timeout, etc., will throw a warning. It's kinda harmless but very misleading. Since no other driver does those checks on insert or update, they've been removed from res_config_mysql. Also, all the logic that actually attempted to ALTER the table to fix the issue has been removed. With the move to alembic, the auto-alter functionality is not only unnecessary, it's also dangerous. The other issue is that res_config_mysql calls the mysql_insert_id function inside store_mysql. Presumably the intention was to return the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE IS NON_PORTABLE AND MAY CHANGE. That value is then returned to config realtime as the number of rows inserted. Guess what? The value changed. It now only returns the number of rows inserted if there's an auto increment column on the table, which ps_contacts doesn't have. Otherwise it returns 0. So now, the insert worked but we tell config realtime and sorcery that no rows were inserted. That call to mysql_insert_id was removed and we now always return 1 if the insert succeeded. We're only inserting 1 row at a time anyway. If the insert fails, we still return -1. ASTERISK-26362 #close Reported-by: Carlos Chavez Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4 2016-08-12 11:22 +0000 [dd6fc1bb7d] frahaase * Binaural synthesis (confbridge): Adds libfftw3 as dependency. Adds libfftw3 to the build chain that is is going to be used for binaural synthesis by bridge_softmix. ASTERISK-26292 Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b 2016-09-29 13:08 +0000 [20c3dba39e] Torrey Searle * res_fax: Fix a tight race condition causing fax to crash in audio fallback When T.38 gets rejected and G711 failback occurs there is a period of time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set, leading to a crash. Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982 2016-10-06 09:58 +0000 [86e8716952] George Joseph * app_dial: Add the "Q" option to set the cause on unanswered channels The "Q" option will set the cause on the unanswered channels when another channel answers. It overrides the default of ANSWERED_ELSEWHERE. NOTE: chan_sip does not support setting the cause on a CANCEL to anything other than ANSWERED_ELSEWHERE. ASTERISK-26446 #close Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47 2016-10-11 06:55 +0000 [4f7f8a7e95] Alexander Traud * chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. In the SIP channel driver chan_sip, auto_comedia was expected to be used in tandem with auto_force_rport. Or stated differently: Only when auto_force_rport was chosen (the default), auto_comedia worked. This change allows auto_comedia to be set independently of the state of (auto_)force_rport. For example, nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments when IPv6 clients are behind a Firewall. ASTERISK-26457 #close Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2 2016-10-10 16:59 +0000 [17031f12fe] Badalyan Vyacheslav * vector: After remove element recheck index Small fix. It is necessary to double-check the index that we just removed because there is a new element. ASTERISK-26453 #close Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7 2016-09-29 12:52 +0000 [cc269766b8] Torrey Searle * res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge If a bridge switched to P2P when a DTMF was in progress it was possible for the DTMF to continue being sent indefinitely. Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29 2016-10-09 21:28 +0000 [fafdde322c] Corey Farrell * logger: Prevent output of verbose messages initiated from rasterisk. Remote asterisk consoles should only display verbose log messages created by the daemon. The first patch for ASTERISK-26410 caused a couple verbose messages to be printed when the rasterisk process ended. ASTERISK-26410 Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a 2016-10-04 20:46 +0000 [7af7490e42] Michael Walton * audiohooks: Remove redundant codec translations when using audiohooks The main frame read and write handlers in main/channel.c don't use the optimum placement in the processing flow for calling audiohooks callbacks, as far as codec translation is concerned. This change places the audiohooks callback code: * After the channel read translation if the frame is not linear before the translation, thereby increasing the chance that the frame is linear as required by audiohooks * Before the channel write translation if the frame is linear at this point This prevents the audiohooks code from instantiating additional translation paths to/from linear where a linear frame format is already available, saving valuable CPU cycles ASTERISK-26419 Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f 2016-10-10 10:59 +0000 [3ab7fae96b] Badalyan Vyacheslav * res_pjsip_config_wizard: Memory leak in module_unload Fixed a memory leak. It removes only the first element. Added a useful feature in vector.h to remove all items under the CMP through a callback function / macro. ASTERISK-26453 #close Change-Id: I84508353463456d2495678f125738e20052da950 2016-09-29 12:45 +0000 [9f62feca60] Ludovic Gasc (GMLudo) * res_calendar: Add support for fetching calendars when reloading We use a lot res_calendar, we are very happy with that, especially because you use libical, the almost alone opensource library that supports really ical format with all types of recurrency. Nevertheless, some features are missed for our business use cases. This first patch adds a new option in calendar.conf: fetch_again_at_reload. Be my guest for a better name. If it's true, when you'll launch "module reload res_calendar.so", Asterisk will download again the calendar. The business use case is that we have a WebUI with a scheduler planner, we know when the calendars are modified. For now, we need to define 1 minute of timeout to have a chance that our user doesn't wait too long between the modification and the real test. But it generates a lot of useless HTTP traffic. ASTERISK-26422 #close Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077 2016-10-09 21:53 +0000 [ca2f3e5b99] Badalyan Vyacheslav * cel_odbc: Fix memory leak on module unload Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715 2016-10-03 11:30 +0000 [5fb848eebd] George Joseph * bundled_pjproject: Add tests for programs used by the Makefile, et al. Added tests for bzip2, tar, patch, sed and nm to configure.ac. Set DOWNLOAD_TO_STDOUT to a working command line regardless of whether the download program is wget, curl or fetch. Added a 'configure.m4' file to the third-party directory which takes care of calling any third-party project setup. Had to move some pjproject_bundled stuff up in configure.ac so it was called before the third-party configure macro. The pjproject tarball is now downloaded to the externals_cache_dir if it was specified on the ./configure command line Removed regeneration of the pjproject aconfigure file. It was only needed for an old patch that no longer applies. Converted the tests for symbols to explicit tests since we know that they're now available in the bundled version. Saves a little time during configure. ASTERISK-26416 #close Reported-by: Corey Farrell Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b (cherry picked from commit e6b0053d7561032b7adbf6f3afaecf30f5046605) (cherry picked from commit a0d02f38322c2c4d7743504003fd376d32a133db) 2016-10-09 18:54 +0000 [73f75c246b] Joshua Colp * Revert "Packet-Loss Concealment (PLC) for supporting codecs." This change introduced some fax test failures that have not yet been addressed. So this is not forgotten I'm submitting a change which reverts it. This reverts: d56fc3b36b7bb59b5506129b9895b6c3341350c9. ASTERISK-25629 Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5 2016-10-05 14:53 +0000 [c5e8f50169] George Joseph * pjproject_bundled: Add MALLOC_DEBUG capability pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0 2016-10-04 16:59 +0000 [442b597929] George Joseph * alembic: Allow cdr, config and voicemail to exist in the same schema cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_'. Regardless, the script then continues with the migration using 'alembic_version_' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898 2016-10-05 04:25 +0000 [c4268ec734] Alexander Traud * chan_sip: Honor support of Symmetric Response (rport) for SIP requests. In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6 2016-08-12 11:22 +0000 [c455823657] frahaase * Binaural synthesis (confbridge): interleaved two-channel audio. Asterisk only supports mono audio at the moment. This patch adds interleaved two-channel audio to Asterisk's channels. ASTERISK-26292 Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a 2016-09-16 18:54 +0000 [2a03575c30] Corey Farrell * astobj2: Add backtrace to log_bad_ao2. * Compile __ast_assert_failed unconditionally. * Use __ast_assert_failed to log messages from log_bad_ao2 * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751 2016-09-30 16:29 +0000 [79532bca75] Rodrigo Ramírez Norambuena * Add text of cdr directory into README.md for ast-db-manage Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636 2016-09-09 12:38 +0000 [806d08b675] Etienne Lessard * app_queue: Update dynamic members ringinuse on reload. Previously, when reloading the members of a queue, the members added statically (i.e. defined in queues.conf) would see their "ringinuse" value updated but not the members added dynamically. This change makes dynamic members ringuse value to be updated on reload. Note that it's impossible to add a dynamic member with a specific ringinuse value. For both static and dynamic members, the ringinuse value can always be changed later on with command like "queue set ringinuse" or with the AMI action "QueueMemberRingInUse". So it's possible this commit could break a user workflow if he was changing the ringinuse value of dynamic members via such commands and was also relying on the fact that a queue reload would not update the dynamic members ringinuse value. ASTERISK-26330 Change-Id: I3745cc9a06ba7e02c399636f1ee9e58c04081f3f 2016-09-29 14:02 +0000 [d31ffb421c] Kevin Harwell * Remove "format_ogg_opus: New format" This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c. ASTERISK-26426 #close Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5 2016-09-19 04:46 +0000 [8c5c95ad89] Corey Farrell * core: Remove ABI effects of LOW_MEMORY. This allows asterisk to compiled with LOW_MEMORY to load modules built without LOW_MEMORY. ASTERISK-26398 #close Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d 2016-09-27 16:10 +0000 [a77ebb2017] George Joseph * download_externals: Fix issue with re-install Needed to ignore an xmlstarlet return code for optional element. Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 Reported-by: Dan Jenkins 2016-09-27 15:35 +0000 [2d2a8944be] Corey Farrell * logger: Output early verbose messages to console. Verbose messages should be printed to the console if the sublevel is less than option_verbose. This fix ensures the welcome message with copyright and license are printed at daemon and interactive rasterisk startup. ASTERISK-26410 #close Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03 2016-09-22 09:49 +0000 [c7ef1e0af3] George Joseph * codec_opus: Add download ability to menuselect Updated codecs/codecs.xml to add codec_opus to the external download list. ASTERISK-26409 Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4 (cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c) (cherry picked from commit e9684f3acd0e8def0df582c1505dd39dd3fd1610) 2016-07-23 14:50 +0000 [5cc3c6679f] George Joseph * codec_opus: Replace res_format_attr_opus with the one from codec_opus Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9) 2016-07-23 15:56 +0000 [40aa28131b] George Joseph * format_ogg_opus: New format Add Ogg/Opus playback support. This uses libopusfile in order to be able to read .opus files and play them back. Writing/recording support is not present at this time. ASTERISK-26409 Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955 (cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa) 2016-09-24 19:05 +0000 [43901e9418] George Joseph * build_tools: Add ability to download variants to download_externals Some external packages have multiple variants that apply to different builds of asterisk. The DPMA for instance has a "bundled" variant that needs to be downloaded if asterisk was configured with --with-pjproject-bundled. There are 2 ways to specify variants: If you need the user to make the decision about which variant to download, simply create multiple menuselect "member" entries like so... external xmlstarlet bash no external xmlstarlet bash no Note that the second entry has "-" appended to the name. You can then use the existing menuselect facilities to restrict which members to enable or disable. Youy probably don't want the user to enable multiple at the same time. If you want to hide the details of the variants, the better way to do it is to create 1 member with "variant" elements. external xmlstarlet bash no The condition must be a bash expression suitable for use with an "if" statement. Any environment variable can be used plus those available in makeopts. In this case, if asterisk was configured with --with-pjproject-bundled the bundled variant will be automatically downloaded. Otherwise the normal version will be downloaded. Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e 2016-09-23 09:54 +0000 [5dd99465d3] Alexander Traud * chan_sip: Resolve externhost not to IPv6; instead go for IPv4. For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac 2016-09-20 09:42 +0000 [d425971009] George Joseph * chan_sip: Address runaway when realtime peers subscribe to mailboxes Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0 2016-09-22 01:40 +0000 [18a8ca06eb] Aaron An * channels/chan_pjsip: fix HANGUPCAUSE function bug. HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61 2016-09-21 14:24 +0000 [a805d779e8] Joshua Colp * core: Ensure presencestate subtype and message are NULL. When retrieving presence state information there is no guarantee that the subtype and message passed in are set to NULL. This change ensures they are. ASTERISK-26397 #close Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86 2016-09-21 10:48 +0000 [077caf566e] Joshua Colp * res_odbc: Make pooling option deprecation notice more useful. This changes the notice for the deprecation of the old pooling options to point to the new option for doing pooling. This gives a clearer direction as to what to look into. ASTERISK-26389 #close Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10 2016-09-21 08:46 +0000 [78b6190a11] Joshua Colp * odbc: Remove options that are no longer applicable. The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6 2016-08-16 15:21 +0000 [923edf2596] Corey Farrell * logger: Simplify ast_callid handling code. Routines responsible for managing ast_callid's are overly complicated. This is left-over code from when ast_callid was an AO2 object. Now that it is an integer the code can be reduced. ast_callid handler code no longer prints it's own error message upon failure to allocate threadstorage as ast_calloc would have already printed a message. Debug messages that were printed when TEST_FRAMEWORK was enabled have been also been removed. Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e 2016-09-20 15:17 +0000 [5cb905a227] Corey Farrell * core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4 2016-09-20 09:22 +0000 [00f1d05d34] Corey Farrell * logger: Always enable verbose for console channel. Previous versions of Asterisk did not require verbose to be specified in logger.conf for the console channel, if it was requested by command line or asterisk.conf it just worked. This change causes Asterisk to always enable verbose in the console channel level mask. Verbose is displayed on consoles if requested by command line, option_verbose or 'core set verbose'. This also delays initialization of the logger until after threadstorage is initialized. Initializing too early can cause messages to be printed multiple times to the console (stdout). ASTERISK-26391 #close Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04 2016-09-20 10:16 +0000 [74f562a8e2] Corey Farrell * logger: Fix default console settings. When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3 2016-09-19 14:21 +0000 [0bc9912739] Walter Doekes * asterisk.c: Non-root users also get the astcanary after core restart. Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55 2016-09-19 09:40 +0000 [bffaf46690] Walter Doekes * asterisk.c: When astcanary dies on linux, reset priority on all threads. Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39 2016-09-12 18:00 +0000 [2820b13393] Richard Mudgett * res_config_odbc.c: Fix buffer size limitation creating invalid SQL. Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae 2016-09-14 06:53 +0000 [0376af9519] Joshua Colp * rtp: Only accept the first payload for a format in SDP. When receiving an SDP offer with multiple payloads for the same format we would generate an answer with the first payload, but during the payload crossover operation (to set the payloads for receiving) we would remove all payloads but the last. This would result in incoming traffic being matched against the wrong format and outgoing traffic being sent using the wrong payload. This change makes it so that once a format has a payload number put into the mapping all subsequent ones are ignored. This ensures there is only ever one payload in the mapping and that it is the payload placed into the answer SDP. ASTERISK-26365 #close Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60 2016-09-14 08:42 +0000 [9d894ee0a1] Joshua Colp * res_pjsip_multihomed: Change Contact port to listening port. The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab 2016-09-07 14:48 +0000 [47c527df0a] George Joseph * pjproject_bundled: Prevent SERVFAIL from marking name server bad A name server that returns "Server Failure" is indicating only that the server couldn't process that particular request. We should NOT assume that the name server is incapable of serving other requests. Here's the scenario we've been encountering... * 2 local name servers configured in resolv.conf. * An OPTIONS request causes a request for A and AAAA records to go out to both nameservers. * The A responses both come back successfully resolved. * Because of an issue at some upstream nameserver, the AAAA responses for that particular query come back as "SERVFAIL" from both local name servers. * Both local servers are marked as bad and no further queries can be sent until the 60 second ttl expires. Only previously cached results can be used. * In this case, 60 seconds is just enough time for another OPTIONS request to go out to the same host so the cycle repeats. We could set the bad ttl really low but that also affects REFUSED and NOTAUTH which probably DO signal a real server issue. Besides, even a really low bad ttl would be an issue on a pbx. Although we use our own resolver in 14 and master and don't have this issue there, Teluu has merged this patch upstream so it's appropriate to cherry-pick to 14 and master to keep pjproject consistent. Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0 2016-09-12 07:37 +0000 [d3ddf4b0fd] Tzafrir Cohen * cdr_mysql: fix UTC support * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778 2016-06-27 14:26 +0000 [07b95f7c65] Tzafrir Cohen * sd_notify (systemd status notifications) support sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 2016-09-09 06:35 +0000 [bc81765bb4] Timo Teräs * Fix showing of swap details when sysinfo() is available If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant || defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca 2016-09-14 07:59 +0000 [89764f7ae9] Joshua Colp * rtp: Preserve timestamps on video frames. Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd 2016-09-14 09:51 +0000 [5f54ac3a80] Joshua Colp * res_pjsip_transport_management: Convert time in log message to seconds. ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc 2016-09-13 05:34 +0000 [6ba68b486e] Steve Davies * chan_sip: Fix session timeout on retransmit of non-UDP packets Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204 2016-09-13 06:08 +0000 [e3487b9360] Joshua Colp * res_pjsip: Don't assume a request will have any addresses. When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c 2016-09-12 12:25 +0000 [7d7b23f04f] Richard Mudgett * app_queue: Fix CLI "queue show" and AMI Queues action output truncation. The output of CLI "queue show" and AMI Queues action is truncated and "failed to extend from 240 to 327" messages are generated if the queue member and interface names are lengthy. * Increase the string buffer size from 240 to 512 in order to accommodate for more information fields added to the output since v1.8. ASTERISK-26360 #close Reported by: Richard Mudgett Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d 2016-09-12 03:28 +0000 [740292e6ae] Walter Doekes * chan_sip: Allow target refresh (Contact update) on re-INVITE. Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from *us* the Contact could get updated, but after re-INVITEs from the *peer*, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435 2016-08-31 15:22 +0000 [82ec58aa91] Richard Mudgett * sip_to_pjsip.py: Map legacy_useroption_parsing. Map the sip.conf general section legacy_useroption_parsing to the new pjsip.conf global ignore_uri_user_options. ASTERISK-26316 Reported by: Kevin Harwell Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc 2016-08-29 18:08 +0000 [ba362822f3] Richard Mudgett * res_pjsip: Add ignore_uri_user_options option. This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62 2016-09-09 06:26 +0000 [56caf5402c] Walter Doekes * contrib: Let safe_asterisk script continue without /dev/tty9. If you use the safe_asterisk script, it uses hardcoded defaults before running configurable values from /etc/asterisk/startup.d. The hardcoded default has TTY=9. Some containerized environments don't have such a TTY, and safe_asterisk would stop. The custom configuration from /etc/asterisk/startup.d/* isn't read until after it stopped, so changing TTY in a custom config did not help. This changeset changes safe_asterisk to continue if the TTY setting was untouched and /dev/tty9 and /dev/vc/9 aren't found. Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc 2016-09-09 05:39 +0000 [901e612739] Joshua Colp * res_pjsip: Only invoke unidentified endpoint logic when unidentified. The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f 2016-08-29 22:26 +0000 [2a50c29101] Aaron An * res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762 2016-08-16 15:34 +0000 [28b2aeba0b] Mark Michelson * res_pjsip: Do not crash on ACKs from unknown endpoints. The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 2016-08-23 06:35 +0000 [82a3d659dc] Joshua Colp * chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 2016-09-06 11:46 +0000 [f369dbb705] Richard Mudgett * res_pjsip_messaging.c: Misc cleanups and fixes. * Eliminated RAII_VAR in get_outbound_endpoint(). * Simplify update_to() coding. However, this function can only be a NoOp because the To string can only be a URI and not a name-address formatted string. * Simplify update_from() coding. Also fixed a code path modifying the from string when the caller could still want to use the original string. * Fixed msg_data_create() incompletely removing the "pjsip:" to then add back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" case because it left the colon after pjsip in the string. Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db 2016-09-07 16:00 +0000 [2e5da0c715] Joshua Colp * res_pjsip: Allow global headers to be overridden. Currently when you add global headers from the dialplan both the header in the dialplan and the globally configured header are added to the resulting SIP INVITE. This change makes it so the headers in the dialplan take precedence and are the only ones added. Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad 2016-08-10 15:14 +0000 [ac02bbd9a0] Mark Michelson * ConfBridge: Make some announcements asynchronous. Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a 2016-07-19 09:41 +0000 [7a12355dbd] Alexander Traud * chan_sip: Allow Preferred sRTP. Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd 2016-09-07 05:59 +0000 [baa7dba180] Joshua Colp * res_resolver_unbound: Fix config documentation. The code was referencing the config section as 'globals' instead of 'general'. This change swaps it over to 'general'. Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe 2016-09-06 15:25 +0000 [e769c19a31] Matt Jordan * res/res_stasis_playback: Cancel the entire playlist when a stop occurs Prior to this patch, a stop issued by a delete of a Playback resource (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop the current media URI playing. Subsequent URIs specified by a playback operation would then proceed on, even though we had just indicated to the User that the Playback was finished *and* after they had just 'deleted' the resource. Whoops. This patch corrects it by bailing out of the sequence of URIs to play if one of them is terminated with an AST_CONTROL_STREAM_STOP indication. ASTERISK-26341 #close Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42 2016-08-01 20:55 +0000 [6caf6bcdad] George Joseph * build: Add download capability for external packages The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a 2016-08-18 14:45 +0000 [7bb7f7b9d5] Alexei Gradinari * res_pjsip_session: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7 2016-09-06 02:41 +0000 [d80b28560c] Walter Doekes * chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2 2016-09-03 16:04 +0000 [730cb3b0b7] Matt Jordan * apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d 2016-09-03 16:02 +0000 [6e1a3b924e] Matt Jordan * apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358 2016-08-30 16:40 +0000 [68c7694abb] Richard Mudgett * res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09 2016-08-29 18:06 +0000 [35ce4d25c7] Richard Mudgett * pjsip_configuration.c: Ignore repeated identify by methods. Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838 2016-08-30 17:26 +0000 [c1e438fdf7] Richard Mudgett * config_global.c: Comments and a default expression adjustment. Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3 2016-08-31 15:14 +0000 [edcf09e47c] Richard Mudgett * sip_to_pjsip.py: Map canreinvite as directmedia alias. Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2 2016-08-31 15:37 +0000 [47336a0bdd] Richard Mudgett * sip_to_pjsip.py: Fix typo converting outboundproxy registration. Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15 2016-08-31 15:13 +0000 [dba02575fc] Richard Mudgett * sip_to_pjsip.py: Fix comment typo and tabs. Change-Id: If35174614545727817d329c60ba4456c028941b5 2016-08-31 15:56 +0000 [4aaa27e532] Richard Mudgett * Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6 2016-09-02 11:36 +0000 [c3b965a2c0] Richard Mudgett * format_cap.c: Fix CLI "core show channeltype Surrogate" crash. * Make ast_format_cap_get_names() NULL tolerant. ASTERISK-26331 #close Reported by: CGI.NET Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3 2016-08-26 17:22 +0000 [e875e1c12a] Corey Farrell * sorcery: Create function ast_sorcery_lockable_alloc. Create an alternative to ast_sorcery_generic_alloc which uses astobj2 shared locking. Use this new method for the 'struct ast_sip_aor' allocator. Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f 2016-08-18 13:28 +0000 [131baf70d6] Corey Farrell * named_locks: Use ao2_weakproxy to deal with cleanup from container. This allows standard ao2 functions to be used to release references to an ast_named_lock. This change can cause less frequent locking of the global named_locks container. The container is no longer locked when a named_lock reference is being release except when this causes the named_lock to be destroyed. Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6 2016-08-26 13:18 +0000 [0c5b6e9ff5] Corey Farrell * astobj2: Support using a separate object for locking. Create ao2_alloc_with_lockobj function to support shared locking. Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80 2016-08-31 12:23 +0000 [48fd4c815c] Michael Kuron * app_mp3: Use correct buffer size and the same sample rate as the channel Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0 2016-08-31 05:33 +0000 [91993ebaa5] Jean Aunis * resource_channels.c: add hangup reason "answered_elsewhere". In ARI, the channels API allows to hangup a channel with a hangup reason. This commit adds a new reason "answered_elsewhere". When using a SIP channel, this will eventually allow Asterisk to add a proper "Reason" header to a CANCEL message. ASTERISK-26321 Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d 2016-08-26 10:39 +0000 [faf9bdebb7] Alexei Gradinari * res_pjsip: qualify/unqualify added/deleted realtime endpoints If the PJSIP endpoint's AOR with the permanent contact was deleted from the realtime storage the res_pjsip module continues trying to qualify this contact. The error 'Unable to find an endpoint to qualify contact' appeares every 'qualify_frequency' seconds. This patch deletes this contact in this case. The PJSIP endpoint's AOR with the permanent contact is never qualified if it is added to realtime storage after asterisk started. This patch adds qualifying for the AOR's permanent contacts on the first handling of this AOR. ASTERISK-26319 #close Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe 2016-08-22 17:08 +0000 [c98a047ee6] Mark Michelson * res_pjsip: Default endpoints to the "offline" status. A recent change attempted to optimize startup by not updating contact status. Instead, code responsible for qualifying contacts updates the status as it becomes known. The code even accounts for contacts/AORs that are not set to be qualified. The problem, though, is when there are no contacts associated with an endpoint. A common case is when an endpoint is set to register its contacts but has not done so yet. In this case, prior to registration, the endpoint's device state will appear to be "not in use" and hints associated with that device will appear to be "idle". In actuality, the device state and hint should both appear as "unavailable". The reason for the failure is that the optimization change made all persistent endpoint states set to "unknown". The fix here is to change the hard-coded "unknown" to be "offline" instead. The default state will be offline until the qualifying code determines that the contact is actually online. This way, if there are no contacts at all, then the state stays as offline, and device state and hints appear correctly. ASTERISK-26269 #close Reported by nappsoft Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a 2016-08-29 07:07 +0000 [5e0758575c] Etienne Lessard * pbx.c: Prevent infinite recursion in manager_show_dialplan_helper. Previously, if context A was including context B and context B was including context A, i.e. if there was a circular dependency between contexts, then calling manager_show_dialplan_helper could lead to an infinite recursion, resulting in a crash. This commit applies the same solution as the one implemented in the show_dialplan_helper function. The manager_show_dialplan_helper and show_dialplan_helper functions contain lots of code in common, but the former was missing the "infinite recursion avoidance" code. ASTERISK-26226 #close Change-Id: I1aea85133c21787226f4f8442253a93000aa0897 2016-08-25 07:06 +0000 [c21e6764f1] Joshua Colp * app_queue: Ensure member is removed from pending when hanging up. When dialing channels it is possible that they may not ever leave the not in use state (Local channels in particular) by the time we cancel them. If this occurs but we know they were dialed we explicitly remove them from the pending members container so that subsequent call attempts occur. ASTERISK-26299 #close Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65 2016-08-26 14:34 +0000 [a7487e9261] George Joseph * pjproject_bundled: Disable srtp use by pjmedia The reason for the disable is that while Asterisk works fine with older libsrtp versions, newer versions of pjproject won't compile with them. Debian 6 for instance, has libsrtp 1.4.4 which is older than what pjproject is expecting. We don't use most of pjmedia but we DO use it for SDP negotiation. Luckily disabling srtp in pjmedia doesn't interfere with it's ability to negitiate a secure channel. The proper crypto attributes are negotiated in both directions. ASTERISK-26279 #close Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2 2016-08-26 08:41 +0000 [858fa5eb2c] Alexander Traud * channel: No hung-up on failing security requirements. In your Diaplan, if you specify same => n,Set(CHANNEL(secure_bridge_media)=1) same => n,Set(CHANNEL(secure_bridge_signaling)=1) only the SIP channel driver chan_sip supports this. All other channels drivers like res_pjsip fail. In case of failure, the original sRTP source code released the whole channel, even if not hung-up, yet. This change does not release the channel but instead hangs-up the channel. ASTERISK-26306 Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db 2016-08-20 09:04 +0000 [f35501b8c9] Alexander Traud * sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. When using the migration script sip_to_pjsip.py, and your sip.conf is configured with bindaddr=::, two transports are written to pjsip.conf, one for 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface like in chan_sip. Furthermore, the script internal functions "build_host" and "split_hostport" did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change makes sure, even such addresses are parsed correctly. ASTERISK-26309 Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48 2016-08-04 20:11 +0000 [ea929d766d] Richard Mudgett * res_pjsip: Cache global config options. We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7 2016-08-23 11:02 +0000 [5eb6cb969f] Richard Mudgett * res_fax: Fix deadlock in ast_channel_get_t38_state(). ast_channel_get_t38_state() calls ast_channel_queryoption() with AST_OPTION_T38_STATE. If the passed in channel is a local channel then a deadlock can happen if a channel lock is held when called. * Made ast_channel_get_t38_state() callers not hold a channel lock before calling. * Update ast_channel_get_t38_state() doxygen to note that no channel locks can be held when calling the function. ASTERISK-26203 #close Reported by: Etienne Lessard ASTERISK-24822 #close Reported by: David Brillert ASTERISK-22732 #close Reported by: Richard Mudgett Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214 2016-08-23 10:39 +0000 [277a2d667a] Richard Mudgett * res_fax: Fix deadlock setting FAXMODE channel variable. ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1 2016-08-22 12:31 +0000 [edca14c8a5] Richard Mudgett * res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407 2016-08-23 11:16 +0000 [141cd42880] Richard Mudgett * res_fax.c: Add chan locked precondition comments. Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7 2016-08-23 10:42 +0000 [b86771d1bf] Richard Mudgett * ast_framehook_detach() must be called with the channel locked. The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584 2016-08-22 15:01 +0000 [5744f434f0] Richard Mudgett * ast_framehook_attach() must be called with the channel locked. The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438 2016-08-17 02:51 +0000 [93b7533d74] chris de rock * app_macro: Consider '~~s~~' as a macro start extension. As described in issue ASTERISK-26282 the AEL parser creates macros with extension '~~s~~'. app_macro searches only for extension 's' so the created extension cannot be found. with this patch app_macro searches for both extensions and performs the right extension. ASTERISK-26282 #close Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb 2016-08-24 04:44 +0000 [d2e03c252d] Eugene * chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820 Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth is not supported in IAX2 protocol. Please refer to section 8.6.13 of RFC 5456. But plaintext auth is still supported by Asterisk implementation of IAX2. This support should be dropped. Patch, based on asterisk-dev discussion, adds deprecation warning on startup if 'auth' is set to 'plaintext', changes default values of 'auth' from 'md5, plaintext' to 'md5'. Patch is safe in terms of backwards compatibility, will work even if remote peers have auth=plaintext and we have defaults. auth=plaintext setting will remain deprecated in Asterisk 14 and 15, and IAX2 plaintext support will be removed in Asterisk 16. ASTERISK-22820 #close Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf 2016-08-24 14:42 +0000 [e40aa40aca] George Joseph * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362 2016-07-19 13:14 +0000 [2e79f52d71] Alexander Traud * codecs: Add Codec 2 mode 2400. ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6 2016-08-10 15:14 +0000 [ded22c712a] Mark Michelson * ConfBridge: Rework announcer channel methodology NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0 2016-08-23 05:54 +0000 [065d810d3f] Joshua Colp * Revert "ConfBridge: Rework announcer channel methodology" This reverts commit 5aa877305223faab5a1119276a934893ab9dc138. Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491 2016-08-19 10:21 +0000 [41ee14bfae] Alexei Gradinari * compilation failed with -Werror=maybe-uninitialized The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580 2016-08-20 14:51 +0000 [eb0c9c476f] David M. Lee * res_odbc_transaction: add dep on generic_odbc When res_odbc_transaction depended on res_odbc, it got the generic_odbc headers and libs implicitly. Now that it no longer depends on res_odbc, its dependency on generic_odbc must be explicit. Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911 2016-08-20 11:18 +0000 [12752c64cc] Alexander Traud * pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations. PJProject supports a lot of platforms even Windows, some with different defaults when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. Because of this, if configured with just an IPv6 address/transport, PJProject listens to both IPv4 and IPv6. However, this is not supported by the PJProject team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only server which accepts incoming connections on IPv4. If you try to configure two transports, one with IPv4 and one with IPv6 on the same interface, as expected by the PJProject team, the IPv4 transport is not able to bind because the IPv6 transport listens to both already. One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. Then, you are able to configure two transports, one for each IP version on the same interface. That way, you get a server which works with IPv4 clients and IPv6 clients at the same time over the same interface. Here, this change sets this parameter directly within PJProject to match the expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack servers out of the box like in chan_sip. This change was accepted by the PJProject team as and is expected to arrive in the next version, PJProject 2.6.0. Until then, this change is incorporated in the bundled PJProject of Asterisk. ASTERISK-26309 Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae 2016-08-19 18:19 +0000 [55ccdf93c3] Corey Farrell * Fix checks for allocation debugging. MALLOC_DEBUG should not be used to check if debugging is actually enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it is active. Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53 2016-08-19 14:09 +0000 [8061d9f66f] Corey Farrell * Fix naming mismatch of allocator functions. Allocator functions that take file/line/func parameters are prefixed with single-underscore when MALLOC_DEBUG is not defined, double-underscore when it is defined. This change updates all allocators that accept file/line/func to have the same prototype in either ABI mode. The parameter order of __ast_vasprintf and __ast_asprintf in utils.h have been changed to match that of astmm.h. End-use allocator macro's have been removed from astmm.h and moved to an unconditional part of utils.h. Change-Id: I823bb6ce2b5675b3a4735948f10a3b420e9a023a 2016-08-17 08:10 +0000 [c1b6a79686] Torrey Searle * res_ari: Add http prefix to generated docs updated the uri handler to include the url prefix of the http server this enables res_ari to add it to the uris when generating docs Change-Id: I279335a2625261a8492206c37219698f42591c2e (cherry picked from commit 6f448f32fe9b7379e2630fab7b06205f901f2ded) 2016-08-19 03:59 +0000 [02a82f758e] Alexander Traud * sip_to_pjsip: Add cert_file. When using the migration script sip_to_pjsip.py, cert_file was not migrated to pjsip.conf. A previous change regarding this contained a copy/paste error. ASTERISK-22374 Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b 2016-08-18 09:21 +0000 [1a9555f036] Alexander Traud * sip.conf: tlsclientmethod is using sslv23 as default. When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3 2016-08-16 15:57 +0000 [53a2f7dc88] Kevin Harwell * res_format_attr_g729: Add annexb=no format parameter to SDPs Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0 2016-08-18 17:02 +0000 [7ea133f2ab] Kevin Harwell * rest-api: Swagger scripts were not replacing format variable in file brief Given resource paths did not have 'json' substituted in for the '{format}'. For some auto generated documentation/comment strings it resulted in something like the following: "... REST handler for /api-docs/sounds.{format}" This patch makes sure the resource api's path is properly substituted. ASTERISK-25472 #close Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23 2016-08-18 15:15 +0000 [c7ffd6111d] George Joseph * res_odbc: Correct the dependency relationship with res_odbc_transaction The MODULEINFO dependencies between these 2 modules was reversed. res_odbc should depend on res_odbc_transaction, not the other way around. ASTERISK-25984 #close Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f 2016-08-18 12:04 +0000 [966527249e] Kevin Harwell * sip_to_pjsip: Set correct tls transport method A recent update had a copy/paste error where the unused variable 'val' was being passed to the set_value function instead of the 'method' value itself. This patch passes in the right variable. ASTERISK-22374 Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06 2016-08-10 15:14 +0000 [5aa8773052] Mark Michelson * ConfBridge: Rework announcer channel methodology One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5 2016-08-18 08:19 +0000 [e55d1e47aa] Alexander Traud * sip_to_pjsip: Map the TLS method correctly. When using the migration script sip_to_pjsip.py and tlsclientmethod is not set in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is offering/using not just TLSv1.0 but TLSv1.2 as well. ASTERISK-22374 Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f 2016-08-18 08:17 +0000 [da14c439a3] Alexander Traud * sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. When using the migration script sip_to_pjsip.py, no section of type=system or type=general were created. Therefore the keys compactheaders, timerb, timert1, and useragent were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1 2016-08-18 08:16 +0000 [675721a7ab] Alexander Traud * sip_to_pjsip: Map (session-)timers correctly. When using the migration script sip_to_pjsip.py, session-timers=accept and session-timers=refuse were mapped to wrong values. ASTERISK-22374 Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092 2016-08-18 08:15 +0000 [acc5237e91] Alexander Traud * sip_to_pjsip: Write username even without authname. When using the migration script sip_to_pjsip.py, now the (mandatory) username is written to pjsip.conf, even if there was no (optional) authname in the register string in sip.conf. ASTERISK-22374 Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f 2016-08-18 08:14 +0000 [3eb02235f5] Alexander Traud * sip_to_pjsip: Parse register even with transport. When using the migration script sip_to_pjsip.py and the register string started with a transport in sip.conf - like tls://... - register was not parsed correctly and therefore not migrated correctly to pjsip.conf. ASTERISK-22374 Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2 2016-08-18 08:13 +0000 [9907e2b1c1] Alexander Traud * sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit. When using the migration script sip_to_pjsip.py, those keys got missing. These keys might appear several times and the function "merge_value" tried to collect those. However, because these keys have different names in sip.conf and pjsip.conf, "merge_value" was not able to find the new key name in sip.conf. This change lets "merge_value" search with the old key name in sip.conf and write with the new key name in pjsip.conf. ASTERISK-22374 Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2 2016-08-18 08:11 +0000 [c0e0075718] Alexander Traud * sip_to_pjsip: Map externhost/ip to Transports. When using the migration script sip_to_pjsip.py, the externhost or externip of sip.conf were erroneously written to Endpoints instead to Transports. ASTERISK-22374 Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4 2016-08-18 08:04 +0000 [a937c2ccb1] Alexander Traud * sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry. When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and minexpiry were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b 2016-08-18 08:03 +0000 [163cc2d68f] Alexander Traud * sip_to_pjsip: Write media_encryption. When using the migration script sip_to_pjsip.py, encryption=yes got missing and media_encryption=sdes was not written to pjsip.conf, because of a typo. ASTERISK-22374 Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05 2016-08-18 08:02 +0000 [d8b5970749] Alexander Traud * sip_to_pjsip: Write cos and tos. When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got missed, because of a typo. Therefore, cos and tos were not written to pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused by a copy-and-paste error. ASTERISK-22374 Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2 2016-08-18 07:55 +0000 [38491401b5] Alexander Traud * sip_to_pjsip: Add cert_file and ca_list_path. When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825 2016-08-16 15:36 +0000 [534063fd67] George Joseph * res_pjsip: Add contact_user to endpoint contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 2016-08-17 14:13 +0000 [0b4fa65532] Richard Mudgett * res_pjsip_session.c: Fix unbound srv failover tests. Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover functionality if a TCP connection gets disconnected. Under these conditions, session_inv_on_state_changed() gets a PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new transport. Unfortunately, session_inv_on_tsx_state_changed() also gets the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates the session. * Made session_inv_on_tsx_state_changed() complete terminating the session on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still PJSIP_INV_STATE_DISCONNECTED. ASTERISK-26305 #close Reported by: Richard Mudgett Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d 2016-08-11 12:10 +0000 [046069011b] Tzafrir Cohen * followme: initialize all config items on reload Some configuration directives were not initialized on reload, and hence were not reset to default if they were removed from followme.conf. ASTERISK-26288 #close Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150 2016-08-17 06:12 +0000 [57f4e4428a] Alexander Traud * BuildSystem: Detect ca_list_path capabilities in external PJProject. Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the bundled PJProject but not in an external PJProject. Therefore, ca_list_path could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2 is detected again to enable ca_list_path again. ASTERISK-26303 #close Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0 2016-08-16 12:24 +0000 [a5c0cf4922] George Joseph * ari: Add documentation that path parameters are case-sensitive Added to api.wiki.mustache so that the generated object pages have the notation in the table header as well as under each method that has path parameters. ASTERISK-25492 #close Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf 2016-08-15 15:29 +0000 [824a4e84d1] Corey Farrell * Refactor usage pattern of xmldoc info tag. This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea 2016-06-15 17:10 +0000 [957df73301] Evgeniy Tsybra * chan_sip: Fix lastrtprx always updated Packets are read regulary, when there is no data in buffer fr->frametype is AST_FRAME_NULL. There was no check of frametype and lastrtprx always updated and, therefore, rtptimeout did not work at all. ASTERISK-25270 #close Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d 2016-08-10 14:41 +0000 [e85adbd947] Alexei Gradinari * core: Entity ID is not set or invalid The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6 2016-08-04 20:00 +0000 [13450c80ce] Richard Mudgett * res_sorcery_config.c: Cleanup ao2 container usage idioms. Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a 2016-08-04 15:57 +0000 [d526aa5cbe] Richard Mudgett * sorcery.c: Minor optimizations. * Remove some unused parameters from internal functions: sorcery_wizard_create() sorcery_wizard_update() sorcery_wizard_delete() * Created the struct sorcery_observer_invocation ao2 object without a lock since it is not needed in sorcery_observer_invocation_alloc(). * Cleanup generic ao2 container sorcery object id hash, sort, and cmp functions. Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e 2016-08-01 11:04 +0000 [45e143576f] Richard Mudgett * sorcery.c: Tweak some container declaration formatting. * Tweak sorcery_object_type_alloc() formatting. * Tweak ast_sorcery_init() formatting. Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944 2016-08-11 23:30 +0000 [eca3d2698a] Corey Farrell * pbx.c: Additional fixes to ast_context_remove_extension_callerid2. Do not check registrar of the first extension head. We should only check the registrar when we match the priority. Additionally fix a couple calls to strcmp which used the input callerid instead of the clean version ex.cidmatch. ASTERISK-26233 Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1 2016-08-13 22:02 +0000 [9202ca34a8] Matt Jordan * app_dial: Improve documentation * Add some helpful and other embedded paragraph tags * Document some of the lesser known channel variables set by Dial * Add examples for some common Dial uses, along with some more challenging but useful options Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1 2016-08-13 20:16 +0000 [e9fe08ea37] Matt Jordan * manager: Add tags to relate interrelated events/actions together Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4 2016-08-13 20:15 +0000 [a93cd39ac1] Matt Jordan * manager: Add tags to relate Bridge related events,actions, and apps Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85 2016-08-13 20:14 +0000 [d8a7594ffd] Matt Jordan * manager: Add tags to relate AoC events and actions Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e 2016-08-13 20:13 +0000 [243f0cf99a] Matt Jordan * manager: Add tags to relate UserEvent actions/apps/events Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4 2016-08-12 15:53 +0000 [3269cf4c17] Matt Jordan * res_agi: Improve documentation * Groups of AGI commands that have similar functionality now reference each other, and all reference the AGI application for ease of wiki reference. * The documentation for the AGI application has been improved, in particular noting the various AGI types and how they are invoked. * A warning message has been added to DeadAGI, noting that it is deprecated. Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d 2016-08-12 13:53 +0000 [a19f4affe8] Matt Jordan * manager: Add links between related events This patch adds some see-also references between related AMI events. It focuses primarily on those events that are guaranteed to come in pairs, such as DTMFBegin/DTMFEnd, as well as those that occur during the life cycle of an Asterisk channel, such as Newchannel/Hangup. Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3 2016-08-12 11:15 +0000 [ddab42e296] Matt Jordan * func_channel: Reorganize documentation * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b 2016-08-15 07:17 +0000 [922b74169f] Joshua Colp * manager: Clarify that dialplan manipulation actions are under system class. ASTERISK-26246 #close Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657 2016-08-11 22:12 +0000 [9debe1ca26] Corey Farrell * Run mandatory cleanup when startup fails. Errors during startup result in an exit. These error branches should be calling ast_run_atexit(0) to ensure mandatory cleanup is run. ASTERISK-26267 #close Change-Id: If226f2326ae2df7add20040696132214cf2bb680 2016-08-11 11:24 +0000 [d7534e016b] George Joseph * res_pjsip_caller_id: Copy header name to short header name When compact_headers was set, we were sending a zero-length header name for PAI and RPID because we always forced the short header name length to 0. We did this because we cloned the header from "From" and wanted to clear "f" from the sname. By cloning however, we bypass pjproject's automatic logic that sets sname to name if there's no compact form of the header, which there isn't for PAI and RPID. So now we force sname to be the same as name right after we set name. res_pjsip_diversion needed the same treatment for the Diversion header. ASTERISK-26241 #close Change-Id: I633ec139630cd83809aae00336cee4a10077e467 2016-08-11 11:13 +0000 [225fd1003f] Matt Jordan * app_queue: Prevent crash when a call is forwarded to an invalid location When a call forward attempt is made from a Queue member, the current code will hang up the forwarding channel in an off-nominal condition prior to raising the Stasis events informing the rest of Asterisk that the call was forwarded. This will result in a slew of dreaded FRACKs, most likely leading to a crash. This patch modifies the code such that we don't hang up the forwarding channel even in an off-nominal condition until we've safely raised the Stasis messages. ASTERISK-25797 #close Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38 2016-08-11 12:18 +0000 [aeb859dba9] George Joseph * res_pjsip: Fail global load if debug or default_from_user are empty If debug was specified in the global configuration but left blank, the logger would treat it as a wildcard and log all hosts. If default_from_user was empty, a crash would result. The global apply handler now checks for empty strings. ASTERISK-26239 #close ASTERISK-26238 #close Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 2016-08-01 15:07 +0000 [2275494e80] Richard Mudgett * res_pjsip res_pjsip_mwi: Misc fixes and cleanups. * Eliminated RAII_VAR() usage in ast_sip_persistent_endpoint_update_state(). * Added a missing allocation failure check to persistent_endpoint_find_or_create(). * Made persistent_endpoint_find_or_create() create the new object without a lock as it isn't needed. * Cleaned up some ao2 container allocation idioms. * Reordered res_pjsip_mwi.c load_module() and unload_module() Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8 2016-08-04 18:03 +0000 [d4ffbccef6] Richard Mudgett * location.c: Misc fixes and cleanups. * Eliminated most RAII_VAR() usage. * Added several missing allocation failure checks. * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without a lock as it is not needed. Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc 2016-08-11 12:01 +0000 [36b2a40533] George Joseph * autohints: Update CHANGES and extensions.conf.sample Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433 2016-08-02 13:53 +0000 [4a5da6c9b4] Richard Mudgett * taskprocessor.c: Tweak high water checks. * The high water check in ast_taskprocessor_alert_set_levels() would trigger immediately if the new high water level is zero and the queue was empty. * The high water check in taskprocessor_push() was off by one. Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d 2016-08-03 16:24 +0000 [5ba6357be2] Richard Mudgett * res_pjsip: Make aor named lock a mutex. The named aor lock was always being locked for writes so a rwlock adds no benefit and may be slower because rwlocks are biased toward read locking. Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28 2016-07-29 17:41 +0000 [b6e03a5ff3] Richard Mudgett * pjsip_distributor.c: Add missing allocation failure check. Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28 2016-08-11 10:50 +0000 [ac0454f9fa] David M. Lee * Fixed compile flags for non-module libs The non-module libs libasteriskssl.dylib and libasteriskpj.dylib have long been missing the AST_NOT_MODULE compile flag. This was mostly okay, until a recent fix to improve compiler warnings when the AST_MODULE_SELF_SYM is missing broke the build on OS X/macOS/whatever they are calling it these days. Change-Id: I2cb51c890824f001280a5114f2e775f97c163516 2016-08-11 10:50 +0000 [b3c2f1164b] Kevin Harwell * alembic: add auth_username to endpoint's identify_by enum A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 2016-08-08 14:50 +0000 [41aba83ff6] Richard Mudgett * res_srtp: Move SDP SRTP code from the core to res_srtp. A patch made to the master branch (Now the 14 branch) inadvertently made libsrtp a required dependency in order to compile Asterisk. Rather than create dummy defines to substitute for the defines supplied by libsrtp when libsrtp is not available, most of the code in sdp_srtp.c is moved into res_srtp.c. This gets more code out of Asterisk's core that isn't used when SRTP is not available. This also makes another inadvertent required dependency on libsrtp by Asterisk's core unlikely. ASTERISK-26253 #close Reported by: Ben Merrills Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7 2016-08-06 10:57 +0000 [820879415f] Alexei Gradinari * pjsip: Fix deadlock with suspend taskprocessor on masquerade If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b 2016-08-09 12:07 +0000 [d4170df40a] Kevin Harwell * alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b 2016-08-10 11:47 +0000 [8d42ff784d] George Joseph * res_resolver_unbound: Allow compilation with libunbound version < 1.5 libunbound at version 1.4.20 (which CentOS still uses) declared all of their string function parameters as as 'char *'. 1.4.21 changed them all to 'const char *'. Thankfully 1.4.21 also introduced the UNBOUND_VERSION_MAJOR define so configure now checks for that and sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks that and casts away the 'const' if it's not set. Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and Fedora24 (1.5.4). There are a few failing tests to be addressed though. ASTERISK-26283 #close Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148 2016-08-07 09:58 +0000 [c315460abb] Matt Jordan * channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b (cherry picked from commit eec60dd77394f0519895fc6abce3a6f90f6470f1) 2016-08-09 16:19 +0000 [8fe9f1f7f1] Mark Michelson * res_rtp_asterisk: Cache local RTCP address. When an RTCP packet is sent or received, res_rtp_asterisk generates a Stasis event that contains the RTCP report as well as the local and remote addresses that the report pertains to. The addresses are determined using ast_find_ourip(). For the local address, this will typically result in a lookup of the hostname of the server, and then a DNS lookup of that hostname. If you do not have the host in /etc/hosts, then this results in a full DNS lookup, which can potentially block for some time. This is especially problematic when performing RTCP reads, since those are done on the same thread responsible for reading and writing media. This patch addresses the issue by performing a lookup of the local address when RTCP is allocated. We then use this cached local address for the Stasis events when necessary. ASTERISK-26280 #close Reported by Mark Michelson Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556 2016-08-08 19:14 +0000 [827457dca0] Corey Farrell * Produce friendly error when AST_MODULE_SELF_SYM is not defined. Modules must define AST_MODULE_SELF_SYM to be used as the name of a generated function. This produces a friendly error when it's not defined. ASTERISK-26278 #close Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b 2016-08-08 12:53 +0000 [403b63571c] Alexei Gradinari * res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a 2016-08-06 01:37 +0000 [0749f6e6f3] Rodrigo Ramírez Norambuena * res_odbc: Show only when there a fail attempt of connection in CLI When is executed CLI command "odbc show all" every time is show information about variable last_negative_connect. If not there a fail attempt of connection will show date like "1969-12-31 21:00:00". This patch fix there situation for to show only this information when exists a fail attempt before. Change-Id: I7c058b0be6f7642e922de75ee6b82c7276c9f113 2016-08-05 22:06 +0000 [b156a291af] Rodrigo Ramírez Norambuena * cdr_adaptive_odbc: Fix DNSs mixed config quote quoted_identifiers When haved more than once DNSs config and one of their dont set quoted_identifiers and before this is with configurated with quoted_identifiers resulting a truncate statement for a reference null for quote character identifier. This patch initializes quoted flag before build SQL Query Example config for this bugfix case in cdr_adaptive_odbc.conf file [first] connection=asterisk-server1 table=cdr quoted_identifiers=" [second] connection=asterisk-server2 table=cdr [third] connection=asterisk-server3 table=cdr quoted_identifiers=` Change-Id: Ibd95667b468e10d4a19a2b9d88b9934ec7207e1d 2016-08-05 15:34 +0000 [9042ad40f2] Alexei Gradinari * app_voicemail: Add taskprocessor alert level options. On heavy loaded system with IMAP or DB storage, 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. It could happen when the IMAP or DB server dies or is unreachable. It could happen on startup when there are many (thousands) realtime endpoints configured with unsolicited mwi. If the taskprocessor queue reaches the high water level then the alert is triggered and pjsip stops processing new requests until the queue reaches the low water level to clear the alert. This patch adds 2 new 'general' configuration options to tune taskprocessor alert levels: 'tps_queue_high' - Taskprocessor high water alert trigger level. 'tps_queue_low' - Taskprocessor low water clear alert level ASTERISK-26229 #close Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8 2016-08-04 10:16 +0000 [54869e4823] Joshua Colp * res_pjsip_outbound_publish: Use a serializer shutdown group for unload. This change replaces the custom unload process for the outbound publish module with the common serializer shutdown group. ASTERISK-25217 #close Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6 2016-08-04 10:27 +0000 [e711e57106] Kevin Harwell * resource_channels: Sync with ARI stubs This file was out of sync with the current ARI definitions. Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a 2016-08-03 15:41 +0000 [29b0f733a0] Corey Farrell * Add missing checks during startup. This ensures startup is canceled due to allocation failures from the following initializations. * channel.c: ast_channels_init * config_options.c: aco_init ASTERISK-26265 #close Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611 2016-08-03 09:47 +0000 [90b30b21ac] Joshua Colp * astconfigparser: Really handle case where line is simply a comment. The regular expression would match causing the code that handled the line if it was merely a comment to never get executed. Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819 2016-08-01 11:08 +0000 [73bce50ef8] Joshua Colp * sorcery: Use more compatible regex for local expressions. This changes the use of an empty regex for both res_sorcery_config and res_sorcery_memory to "." instead. This is a more compatible regular expression which also works on FreeBSD. ASTERISK-26206 #close Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388 2016-08-02 03:08 +0000 [3ff964c6b6] Alexander Traud * res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058 2016-08-01 16:13 +0000 [f6276441b1] George Joseph * menuselect: Add an opaque "member_data" string to the acceptable xml Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe 2016-07-29 13:13 +0000 [1cd79d6ee5] Mark Michelson * Remove SILK payload mappings from Asterisk core. SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 2016-07-29 04:48 +0000 [a7ae48441f] Joshua Colp * astconfigparser: Handle case where line is simply a comment. Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5 2016-07-28 14:10 +0000 [89a0a1eb45] Corey Farrell * pbx.c: Fix handling of '-' in extension name and callerid This adds a two strings to ast_exten. name to go with exten and cidmatch_display to go with cidmatch. The new fields contain input used to add the extension in the first place. The existing fields now contain stripped input that excludes insignificant spaces and dashes. These stripped fields should always be used for comparisons. The unstripped fields should normally be used for display, but displaying stripped values will not cause runtime errors. Note the actual string is only stored twice if it contains dashes. If no dashes are found then both 'char *' fields point to the same memory. So this change has a minimum effect on memory usage. The existing functions ast_get_extension_name and ast_get_extension_cidmatch return unstripped values as they did before this change. Other similar bugs likely still exist where unstripped extensions are saved outside pbx.c then passed back in. ASTERISK-26233 #close Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f 2016-07-27 17:17 +0000 [68ebf86e2f] Richard Mudgett * pbx.c: Allow dangerous functions when adding a hint to dialplan. We can allow dangerous functions when adding a hint since altering dialplan is itself a privileged activity. Otherwise, we could never execute dangerous functions. ASTERISK-25996 #close Reported by: Andrew Nagy Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba 2016-07-21 10:36 +0000 [b5bc2fdda8] Alexei Gradinari * pjproject: fixed a few bugs This patch fixes the issue in pjsip_tx_data_dec_ref() when tx_data_destroy can be called more than once, and checks if invalid value (e.g. NULL) is passed to. This patch updates array limit checks and docs in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability(). Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40 2016-07-17 18:28 +0000 [b4f1c6380e] George Joseph * pjproject_bundled: Update for pjproject 2.5.5 Add more --disable-* switches to Makefile.rules including --disable-opus which was causing bundled pjproject to fail with "undefined reference" errors in libasteriskpj. Changed PJ_ENABLE_EXTRA_CHECK to 1. Removed 2 obsolete patches and added a new one. The new one was merged by Teluu on 6/27/2016. ASTERISK-26148 #close Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 2016-07-27 10:33 +0000 [feb1a43412] David M. Lee * Portably sscanf tv_usec In a timeval, tv_usec is defined as a suseconds_t, which could be different underlying types on different platforms. Instead of trying to scanf directly into the timeval, scanf into a long int, then copy that into the timeval. Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95 2016-07-27 12:36 +0000 [1d364ac54f] Kevin Harwell * rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c 2016-07-27 09:56 +0000 [8802e55c26] David M. Lee * Replace strdupa with more portable ast_strdupa The strdupa function is a GNU extension, and not widely portable. We have an ast_strdupa function used within Asterisk which is preferred. I pulled the definition up from menuselect.c into the menuselect.h header file so it can be shared across menuselect. Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e 2016-07-21 22:44 +0000 [737471f131] Richard Mudgett * dsp.c: Add fax and DTMF detection unit tests. * Add fax amplitude and frequency sweep tests. * Add DTMF amplitude and twist unit tests. Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7 2016-07-21 11:56 +0000 [a8cd5d255a] Richard Mudgett * dsp.c: Added descriptive comments to Goertzel calculations. * Added doxygen to describe some struct members and what is going on in the code. Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d 2016-07-13 13:48 +0000 [6dfb34cf13] Richard Mudgett * dsp.c: Fix incorrect format reference typo. Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896 2016-07-25 21:18 +0000 [327136088e] Richard Mudgett * dsp.c: Correct DTMF twist dsp.conf documentation. Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae 2016-07-22 04:43 +0000 [1e7168aee0] Joshua Colp * astconfigparser.py: Update with realtime fixes. When configuring SIP URIs in the pjsip.conf file it is necessary to escape the semicolon so the parser does not treat it as a comment. This change allows this to work in the astconfigparser implementation. A secondary bug where some data was lost if a configuration option included a "=" in its value was also fixed. A bug where sections would be considered equal despite being different has also been fixed. Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8 2016-07-21 22:28 +0000 [49461f37b7] Richard Mudgett * dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc 2016-07-24 18:27 +0000 [b4c5dcad01] George Joseph * menuselect: Various menuselect enhancements * Add 'external' as a support level. * Add ability for module directories to add entries to the menu by adding members to the /.xml file. * Expand the description field to 3 lines in the ncurses implementation. * Allow the description field to wrap in the newt implementation. * Add description field to the gtk implementation. Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 2016-07-24 16:51 +0000 [9db420c69d] Joshua Colp * ari: Update version. New functionality has been added so the version has been bumped to one over the 13 version. Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69 2016-07-23 08:51 +0000 [8852a4c3db] George Joseph * asterisk.c: Add auto generation and persistence of UUID Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d 2016-07-22 14:44 +0000 [76781a0964] Mark Michelson * Fix sqlalchemy error regarding identifier length. sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 2016-07-19 06:16 +0000 [9be69c1636] Alexander Traud * chan_sip: Enable Session-Timers for SIP over TCP (and TLS). Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 2016-07-19 13:39 +0000 [8fb807009f] Alexander Traud * codecs: Add iLBC 20. Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests this. ASTERISK-26218 #close ASTERISK-26221 #close Reported by: Aaron Meriwether Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa 2016-07-15 16:16 +0000 [4286a369a1] Richard Mudgett * res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 2016-07-21 22:34 +0000 [68de3a9e51] Corey Farrell * pbx.c: Remove duplicate code. Merge code found in both branches of a conditional in ast_add_extension2_lockopt. The updated code initializes peer_table and peer_label_table of the extension before linking it to the context. Change-Id: Ic759e27cdc9906c6877df41d28ee9c5be8f41c20 2016-07-21 16:35 +0000 [15bf6a87dc] George Joseph * Create Asterisk-14: Update CHANGES and UPGRADE files Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc 2016-07-21 09:05 +0000 [1b4922466b] George Joseph * chan_sip: Prevent deadlock when issuing "sip show channels" sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 2016-07-15 19:28 +0000 [a36a174c4b] Corey Farrell * pbx: Create pbx_sw.c for management of 'struct ast_sw'. This changes context switches from a linked list to a vector, makes 'struct ast_sw' opaque to pbx.c. Although ast_walk_context_switches is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_switches_count (AST_VECTOR_SIZE) * ast_context_switches_get (AST_VECTOR_GET) As with ast_walk_context_switches callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the switches, they have been converted to use the new functions. Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998 2016-07-21 10:28 +0000 [81ea024d93] Alexei Gradinari * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a 2016-07-13 05:24 +0000 [1d2173c7ae] Alexander Traud * res_srtp: Enable AES-256 and AES-GCM. ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b 2016-07-18 22:46 +0000 [8f6e9ffcc6] Corey Farrell * Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 2016-07-19 13:18 +0000 [3d62f317dd] Richard Mudgett * chan_dahdi.c: Fix deadlock potential in fax redirection. The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. ASTERISK-26216 #close Reported by: Richard Mudgett Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa 2016-07-13 18:49 +0000 [db4979fa79] Richard Mudgett * chan_sip.c: Fix deadlock potential in fax redirection. The sip_read() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e 2016-07-13 18:48 +0000 [3db468ea9e] Richard Mudgett * chan_pjsip.c: Fix deadlock potential in fax redirection. The chan_pjsip_cng_tone_detected() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 2016-07-12 17:33 +0000 [9abbea162c] Richard Mudgett * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. The fax_detect_framehook() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. * Made only detach the framehook if we detected a fax and not on other possible frames. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d 2016-07-12 17:24 +0000 [804fbd9c2b] Richard Mudgett * res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 2016-07-18 16:16 +0000 [0d1744e132] Richard Mudgett * chan_dahdi: Add faxdetect_timeout option. The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 2016-07-15 20:44 +0000 [e739888d99] Richard Mudgett * res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d 2016-07-17 07:43 +0000 [d56fc3b36b] Alexander Traud * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ASTERISK-25629 #close Change-Id: I66c0086e6c17764b8141ec60a3e2aaefe088eb78 2016-09-19 14:18 +0000 Asterisk Development Team * asterisk 14.0.0-rc1 Released. 2016-09-19 09:17 +0000 [a23b33576f] Joshua Colp * Release summaries: Add summaries for 14.0.0-rc1 2016-09-19 09:17 +0000 [e11354b864] Joshua Colp * Release summaries: Remove previous versions 2016-09-19 09:17 +0000 [24fac2271a] Joshua Colp * .version: Update for 14.0.0-rc1 2016-09-19 09:17 +0000 [52c101d441] Joshua Colp * .lastclean: Update for 14.0.0-rc1 2016-09-19 09:17 +0000 [edae56dc65] Joshua Colp * realtime: Add database scripts for 14.0.0-rc1 2016-09-14 09:51 +0000 [205e2ea351] Joshua Colp * res_pjsip_transport_management: Convert time in log message to seconds. ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc 2016-09-13 06:08 +0000 [bc085bba24] Joshua Colp * res_pjsip: Don't assume a request will have any addresses. When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c 2016-09-09 05:39 +0000 [9a800b24ac] Joshua Colp * res_pjsip: Only invoke unidentified endpoint logic when unidentified. The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f 2016-08-16 15:34 +0000 [137aa2f13c] Mark Michelson * res_pjsip: Do not crash on ACKs from unknown endpoints. The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 2016-08-23 06:35 +0000 [f877e62cc9] Corey Farrell (license 5909) * chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 2016-09-06 15:25 +0000 [b17ee86148] Matt Jordan * res/res_stasis_playback: Cancel the entire playlist when a stop occurs Prior to this patch, a stop issued by a delete of a Playback resource (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop the current media URI playing. Subsequent URIs specified by a playback operation would then proceed on, even though we had just indicated to the User that the Playback was finished *and* after they had just 'deleted' the resource. Whoops. This patch corrects it by bailing out of the sequence of URIs to play if one of them is terminated with an AST_CONTROL_STREAM_STOP indication. ASTERISK-26341 #close Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42 2016-08-29 12:30 +0000 Asterisk Development Team * asterisk 14.0.0-beta2 Released. 2016-08-29 07:29 +0000 [9cdf44668d] Joshua Colp * Release summaries: Add summaries for 14.0.0-beta2 2016-08-29 07:29 +0000 [73d39f2029] Joshua Colp * Release summaries: Remove previous versions 2016-08-29 07:29 +0000 [e8a97775ee] Joshua Colp * .version: Update for 14.0.0-beta2 2016-08-29 07:29 +0000 [345409825a] Joshua Colp * .lastclean: Update for 14.0.0-beta2 2016-08-29 07:29 +0000 [105c1168f7] Joshua Colp * realtime: Add database scripts for 14.0.0-beta2 2016-08-29 06:31 +0000 [8927b52634] Joshua Colp * alembic: Fix downgrade path. The 3772f8f828da version was referencing a previous version that did not exist in the 14.0 branch. It has been fixed to reference the correct previous version. Change-Id: I004d0fcfdfe1d1bb6f01c6dac2b69f6b1f40ae51 2016-08-11 12:18 +0000 [9a95c6dea3] gtjoseph * res_pjsip: Fail global load if debug or default_from_user are empty If debug was specified in the global configuration but left blank, the logger would treat it as a wildcard and log all hosts. If default_from_user was empty, a crash would result. The global apply handler now checks for empty strings. ASTERISK-26239 #close ASTERISK-26238 #close Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 2016-08-11 11:24 +0000 [aaee8160bc] gtjoseph * res_pjsip_caller_id: Copy header name to short header name When compact_headers was set, we were sending a zero-length header name for PAI and RPID because we always forced the short header name length to 0. We did this because we cloned the header from "From" and wanted to clear "f" from the sname. By cloning however, we bypass pjproject's automatic logic that sets sname to name if there's no compact form of the header, which there isn't for PAI and RPID. So now we force sname to be the same as name right after we set name. res_pjsip_diversion needed the same treatment for the Diversion header. ASTERISK-26241 #close Change-Id: I633ec139630cd83809aae00336cee4a10077e467 2016-08-11 12:01 +0000 [7af0eac02a] gtjoseph * autohints: Update CHANGES and extensions.conf.sample Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433 2016-08-11 10:50 +0000 [ef0bf47bb3] Kevin Harwell * alembic: add auth_username to endpoint's identify_by enum A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 2016-08-08 14:50 +0000 [a1d6b14c40] Richard Mudgett * res_srtp: Move SDP SRTP code from the core to res_srtp. A patch made to the master branch (Now the 14 branch) inadvertently made libsrtp a required dependency in order to compile Asterisk. Rather than create dummy defines to substitute for the defines supplied by libsrtp when libsrtp is not available, most of the code in sdp_srtp.c is moved into res_srtp.c. This gets more code out of Asterisk's core that isn't used when SRTP is not available. This also makes another inadvertent required dependency on libsrtp by Asterisk's core unlikely. ASTERISK-26253 #close Reported by: Ben Merrills Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7 2016-08-09 12:07 +0000 [a783e1e60d] Kevin Harwell * alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b 2016-08-10 11:47 +0000 [9c56f798f6] gtjoseph * res_resolver_unbound: Allow compilation with libunbound version < 1.5 libunbound at version 1.4.20 (which CentOS still uses) declared all of their string function parameters as as 'char *'. 1.4.21 changed them all to 'const char *'. Thankfully 1.4.21 also introduced the UNBOUND_VERSION_MAJOR define so configure now checks for that and sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks that and casts away the 'const' if it's not set. Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and Fedora24 (1.5.4). There are a few failing tests to be addressed though. ASTERISK-26283 #close Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148 2016-08-01 16:13 +0000 [1ad00c1c30] gtjoseph * menuselect: Add an opaque "member_data" string to the acceptable xml Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe 2016-07-17 18:28 +0000 [815b6f72f8] gtjoseph * pjproject_bundled: Update for pjproject 2.5.5 Add more --disable-* switches to Makefile.rules including --disable-opus which was causing bundled pjproject to fail with "undefined reference" errors in libasteriskpj. Changed PJ_ENABLE_EXTRA_CHECK to 1. Removed 2 obsolete patches and added a new one. The new one was merged by Teluu on 6/27/2016. ASTERISK-26148 #close Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 (cherry picked from commit 4cf02b5584ce33bb0a64408c27bf20c19bc4ce13) 2016-07-29 13:13 +0000 [c95b611a73] Mark Michelson * Remove SILK payload mappings from Asterisk core. SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 2016-07-27 12:36 +0000 [bc94ccbcdd] Kevin Harwell * rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c 2016-07-26 23:19 +0000 Asterisk Development Team * asterisk 14.0.0-beta1 Released. 2016-07-26 17:22 +0000 [a7233fbf3e] Mark Michelson * Release summaries: Add summaries for 14.0.0-beta1 2016-07-26 16:24 +0000 [c327430ea0] Mark Michelson * Release summaries: Remove previous versions 2016-07-26 16:24 +0000 [763a18bc9d] Mark Michelson * .version: Update for 14.0.0-beta1 2016-07-26 16:24 +0000 [ce6898bd3c] Mark Michelson * .lastclean: Update for 14.0.0-beta1 2016-07-26 16:24 +0000 [ebc477aa5d] Mark Michelson * realtime: Add database scripts for 14.0.0-beta1 2016-07-26 16:00 +0000 [1838b283aa] Mark Michelson * ChangeLog: Updated for 14.0.0 2016-07-26 15:02 +0000 [f196cf975d] Mark Michelson * Release summaries: Add summaries for 14.0.0 2016-07-26 14:01 +0000 [699a7390eb] Mark Michelson * .version: Update for 14.0.0 2016-07-26 14:01 +0000 [4b17a11d7d] Mark Michelson * .lastclean: Update for 14.0.0 2016-07-26 14:01 +0000 [bb9dcae98c] Mark Michelson * realtime: Add database scripts for 14.0.0 2016-07-24 18:27 +0000 [90f445729d] gtjoseph * menuselect: Various menuselect enhancements * Add 'external' as a support level. * Add ability for module directories to add entries to the menu by adding members to the /.xml file. * Expand the description field to 3 lines in the ncurses implementation. * Allow the description field to wrap in the newt implementation. * Add description field to the gtk implementation. Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 2016-07-24 16:51 +0000 [f75401b1e3] Joshua Colp * ari: Update version. New functionality has been added so the version has been bumped to one over the 13 version. Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69 2016-07-23 08:51 +0000 [58759bd77c] gtjoseph * asterisk.c: Add auto generation and persistence of UUID Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d 2016-07-22 14:44 +0000 [46b4e673ae] Mark Michelson * Fix sqlalchemy error regarding identifier length. sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 2016-07-22 07:01 +0000 [633c34c411] gtjoseph * build_tools: Update make_version for 14 Also remove svn stuff Change-Id: I95d762f7cbbe5eb01117bde8779515d51a0bb06a 2016-07-19 13:39 +0000 [c82f24f36a] Alexander Traud * codecs: Add iLBC 20. Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests this. ASTERISK-26218 #close ASTERISK-26221 #close Reported by: Aaron Meriwether Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa 2016-07-15 16:16 +0000 [6e2e3915c8] Richard Mudgett * res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 2016-07-19 13:18 +0000 [5efb5b38e8] Richard Mudgett * chan_dahdi.c: Fix deadlock potential in fax redirection. The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. ASTERISK-26216 #close Reported by: Richard Mudgett Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa 2016-07-13 18:49 +0000 [a1d36c89e0] Richard Mudgett * chan_sip.c: Fix deadlock potential in fax redirection. The sip_read() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e 2016-07-13 18:48 +0000 [4dfadcb025] Richard Mudgett * chan_pjsip.c: Fix deadlock potential in fax redirection. The chan_pjsip_cng_tone_detected() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 2016-07-12 17:33 +0000 [964ae54ecf] Richard Mudgett * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. The fax_detect_framehook() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. * Made only detach the framehook if we detected a fax and not on other possible frames. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d 2016-07-12 17:24 +0000 [c3462adeb8] Richard Mudgett * res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 2016-07-18 16:16 +0000 [c03e27c1c8] Richard Mudgett * chan_dahdi: Add faxdetect_timeout option. The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 2016-07-15 20:44 +0000 [d11731ac2f] Richard Mudgett * res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d 2016-07-21 10:28 +0000 [56b4112659] Alexei Gradinari * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a 2016-07-21 09:05 +0000 [52cbdf2393] gtjoseph * chan_sip: Prevent deadlock when issuing "sip show channels" sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 2016-07-13 05:24 +0000 [2103ad1fec] Alexander Traud * res_srtp: Enable AES-256 and AES-GCM. ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b 2016-07-18 22:46 +0000 [05cfe1a76e] Corey Farrell * Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 2016-07-15 19:28 +0000 [0c88fb460f] Corey Farrell * pbx: Create pbx_sw.c for management of 'struct ast_sw'. This changes context switches from a linked list to a vector, makes 'struct ast_sw' opaque to pbx.c. Although ast_walk_context_switches is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_switches_count (AST_VECTOR_SIZE) * ast_context_switches_get (AST_VECTOR_GET) As with ast_walk_context_switches callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the switches, they have been converted to use the new functions. Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998 2016-07-19 04:48 +0000 [6fca2b3bf0] Alexander Traud * Makefile: Retain XML Declaration and DTD in docs. Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, the XML Declaration and DTD were overwritten by this. ASTERISK-26212 #close Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd 2016-07-18 18:40 +0000 [cf1188a1be] Corey Farrell * Unit tests: Use AST_TEST_DEFINE in conditional code only. If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686 2016-07-18 09:22 +0000 [e9daa34261] Alexei Gradinari * res_pjsip_mwi: remove unneeded check on endpoint's contacts. The function create_mwi_subscriptions_for_endpoint checks if there is active contacts by retrieving aors and contacts. This function is used to create all unsolicited mwi subscriptions on startup and is used when contact added. In both cases it's not necessary to check if there are contacts. The contacts are needed when asterisk sends mwi. ASTERISK-26200 #close Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa 2016-07-18 05:13 +0000 [cb5e3445be] Alexander Traud * res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. With this change, the initial RTP sequence number is randomly chosen not between 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over counter (ROC) synchronization is not lost for sRTP, when the very first RTP packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 ASTERISK-26207 #close Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464 2016-07-18 04:14 +0000 [6428580e7f] Alexander Traud * Makefile: Suppress echoing of target 'config' again. ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f 2016-07-15 02:59 +0000 [e2e8713b84] Corey Farrell * pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'. This changes context ignore patterns from a linked list to a vector, makes 'struct ast_ignorepat' opaque to pbx.c. Although ast_walk_context_ignorepats is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_ignorepats_count (AST_VECTOR_SIZE) * ast_context_ignorepats_get (AST_VECTOR_GET) As with ast_walk_context_ignorepats callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the ignorepats, they have been converted to use the new functions. Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a 2016-07-14 13:51 +0000 [be36bd7ca5] Corey Farrell * pbx: Create pbx_include.c for management of 'struct ast_include'. This changes context includes from a linked list to a vector, makes 'struct ast_include' opaque to pbx.c. Although ast_walk_context_includes is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_includes_count (AST_VECTOR_SIZE) * ast_context_includes_get (AST_VECTOR_GET) As with ast_walk_context_includes callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the includes, they have been converted to use the new functions. const have been applied where possible to parameters for ast_include functions. Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60 2016-07-14 03:25 +0000 [d3348c51b5] Corey Farrell * features.c: Remove unneeded adsi.h include. adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59 2016-06-30 15:58 +0000 [273052f404] Mark Michelson * Update support for SILK format. This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e 2016-07-14 07:45 +0000 [31967dacdf] Richard Miller (license 5685) * app_queue: Only remove queue member from pending when state changes. It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0 2016-07-14 02:40 +0000 [f3608b50d7] Corey Farrell * pbx: Fix leak of timezone for time based includes. Create include_free to run ast_destroy_timing and ast_free, use that in all places that freed an ast_include structure. This fixes a couple of paths that previously did not run ast_destroy_timing. ASTERISK-26196 #close Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838 2016-07-13 17:45 +0000 [63ac4c9487] Kevin Harwell * translate: explicit format destination not properly set If the destination format's name differed from the codec name then the translator's explict_dst field would be improperly set. In some circumstances it would end up setting it to a newly created format that has the same name as the codec when it actually needed to be the given destination codec. This could cause the translation path to use the wrong format. For instance, if an endpoint had specified 'myulaw' as a format the translator could end up using a 'ulaw' format (with whatever/default settings) instead. If the format attribute settings differed between the two then there may unexpected results during processing. This patch removes the name check when building the translation path. This should make it always set the translator's explicit_dst to the given destination format as long as the sample rate and types match. Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5 2016-07-08 11:46 +0000 [2f26512fd8] Richard Mudgett * stasis_endpoint.c: Fix contactstatus_to_json(). The roundtrip_usec json member is optional. If it isn't present then don't put it into the converted json structure where ast_json_pack() will choke on it. Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0 2016-07-11 10:22 +0000 [bc1ff41be7] Richard Mudgett * pjsip_options.c: Fix container operation. aor_observer_deleted() needs to operate on all contacts found for the deleted AOR instead of only the first one found. This is really only a problem if there is more than one contact for the AOR. Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1 2016-07-11 10:21 +0000 [eabcfeeaa3] Richard Mudgett * pjsip_configuration.c: Misc cleanups. * Fix some whitespace in various routines. * Rename i to iter in persistent_endpoint_update_state(). * Fix off-nominal copy/paste message wording in persistent_endpoint_contact_deleted_observer() Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8 2016-07-13 13:45 +0000 [f73ddde7d4] Corey Farrell * chan_sip: Fix reference leak in mwi_event_cb Cleanup the peer reference when stasis_subscription_final_message is true. Also free peer_name even if peer exists, after reload a new peer_name will be allocated. ASTERISK-26193 #close Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69 2016-07-13 11:30 +0000 [fd54d69feb] Corey Farrell * threadpool: Fix leak in ast_threadpool_serializer_group error path. ast_threadpool_serializer_group leaks a reference to ser when listener is allocated but tps is not. Although listener takes the reference to ser cleanup functions are not run without tps. ASTERISK-26191 #close Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585 2016-06-22 07:13 +0000 [85212f2799] Eugene Voityuk ,Alexander Traud * res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c 2016-06-24 19:55 +0000 [0d487b53b1] Matt Jordan * res/res_pjsip_session: Check for presence of an active negotiator It is possible in a hypothetical situation for a session refresh to be invoked on a PJSIP when the negotiatior on the INVITE session has not yet been established. While this shouldn't occur with existing uses of ast_sip_session_refresh, the crashes that occur due to improperly calling PJSIP functions that expect a non-NULL negotiatior are avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this means that simply checking for the presence of the negotiator before passing it to other PJSIP functions that use it is allowable. As such, this patch adds checks for the presence of the negotiator before calling PJSIP functions that assume it is non-NULL. Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d 2015-10-19 18:55 +0000 [c49833653b] Matt Jordan * res/res_pjsip_pubsub: Add additional debug statements When something very sad and wrong occurs, it's challenging sometimes to figure out why. This patch adds some additional debug statements on off-nominal paths to try and make debugging easier. Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640 2015-10-19 18:55 +0000 [f12311ee69] Matt Jordan * res/res_corosync: Raise a Stasis message on node join/leave events When res_corosync detects that a node leaves or joins, it currently is informed of this via Corosync callbacks. However, there are a few limitations with the information presented: (1) While we have information that Corosync is aware of - such as the Corosync nodeid - that information is really only useful inside of Corosync or res_corosync. There's no way to translate a Corosync nodeid to some other internally useful unique identifier for the Asterisk instance that just joined or left the cluster. (2) While res_corosync is notified of the instance joining or leaving the cluster, it has no mechanism to inform the Asterisk core or other modules of this event. This limits the usefulness of res_corosync as a heartbeat mechanism for other modules. This patch addresses both issues. First, it adds the notion of a cluster discovery message both within the Stasis message bus, as well as the binary event messages that res_corosync uses to transmit data back and forth within the cluster. When Asterisk joins the cluster, it sends a discovery message to the other nodes in the cluster, which correlates the Corosync nodeid along with the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids to Asterisk EIDs, such that it can map changes in cluster state with the Asterisk instance that has that nodeid. Likewise, when an Asterisk instance receives a discovery message from a node in the cluster, it now sends its own discovery message back to the originating node with the local Asterisk EID. This lets Asterisk instances within the cluster build a complete picture of the other Asterisk instances within the cluster. Second, it publishes the discovery messages onto the Stasis message bus. Said messages are published whenever a node joins or leaves the cluster. Interested modules can subscribe for the ast_cluster_discovery_type() message under the ast_system_topic() and be notified when changes in cluster state occur. Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465 2016-07-13 08:57 +0000 [a3f4141f6f] Alexander Traud * BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf. Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. ASTERISK-26046 #close Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7 2016-07-11 20:07 +0000 [886f2cab23] gtjoseph * rest_api/channels: Fix multiple issues with create and dial * We weren't properly subscribing to the channel and it's originator on create. * We weren't doing a publish_dial after calling ast_call on dial. * We weren't calling depart_bridge when a channel left the dial bridge. The first 2 issues were causing events to not be generated and the third was actually causing channels to not get properly destroyed when hung up. Together these 3 issues were causing the new rest_apichannels/create_dial_bridge tests to fail. As a result of the fixes, the cdr state machine had to be slightly tweaked to allow bridge leave events without asserting and the tests themselves had to be updated to account for the channels now cleaning themselves up. Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8 2016-07-11 10:25 +0000 [b85446d039] Richard Mudgett * res_pjsip: Fix statsd regression. The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1 2016-07-10 19:08 +0000 [4ad333bb0e] Joshua Colp * func_odbc: Fix connection deadlock. The func_odbc module was modified to ensure that the previous behavior of using a single database connection was maintained. This was done by getting a single database connection and holding on to it. With the new multiple connection support in res_odbc this will actually starve every other thread from getting access to the database as it also maintains the previous behavior of having only a single database connection. This change disables the func_odbc specific behavior if the res_odbc module is running with only a single database connection active. The connection is only kept for the duration of the request. ASTERISK-26177 #close Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f 2016-07-12 03:50 +0000 [110b01a0bc] Alexander Traud * BuildSystem: Allow own CFLAGS on ./configure. Before this change, make failed with the error Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH when CFLAGS were supplied to the configure script. This was introduced with which disabled BUILD_NATIVE when CFLAGS were supplied. Those who need different -march= values, please, go for ./configure make menuselect.makeopts or make menuselect ./menuselect/menuselect --disable BUILD_NATIVE ASTERISK-25289 #close Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc 2016-07-11 13:42 +0000 [44f16af7cc] Richard Mudgett * ast_expr2: Fix off-nominal memory leak. Thanks to ibercom for pointing out a memory leak that was missed in the earlier patch for the issue. ASTERISK-26119 Reported by: Alexei Gradinari Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71 2016-07-11 10:17 +0000 [8476a9332f] Alexander Traud * install_prereq: Checkout of libSRTP 1.5.x. Since 5th November 2014, the master branch of libSRTP changed the prefix of several member names and is not compatible with the source code in Asterisk anymore. Therefore instead, this change checks out the latest version of the libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as backend. This makes AES-GCM and AES-IN possible. ASTERISK-22131 #close Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6 2016-07-09 13:32 +0000 [ad30d60c69] Corey Farrell * chan_sip: Fix reference leaks in error paths. * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. * build_peer leaks peer on failure to allocate the endpoint. This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed with an unref in the appropriate place. ASTERISK-26184 #close Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12 2016-07-07 12:44 +0000 [7408c51a48] Corey Farrell * REF_DEBUG: Prevent logging of container node objects. Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being recorded to the refs log for the node being replaced. This prevents logging of those unrefs since they would produce errors in refcounter.py. ASTERISK-26181 #close Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4 2016-07-04 16:38 +0000 [c832f100d9] Alexei Gradinari * res_sorcery_realtime: fix bug when successful UPDATE is treated as failed If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. This value should be treated as success. But the function sorcery_realtime_update treats it as failed. This bug was found using stress tests on PJSIP. If there are 2 consecutive SIP REGISTER requests with the same contact data during 1 second then res_pjsip_registrar adds contact location on 1st request and tries to update contact location on 2nd. The update fails and res_pjsip_registrar even removes correct contact location. The test "object_update_uncreated" was removed from test_sorcery_realtime.c because it's now a valid situation. This patch also adds missing debug of extra SQL parameter. ASTERISK-26172 #close Change-Id: I05a7f3051455336c9dda29efc229decf86071303 2016-07-07 10:38 +0000 [302be4809a] Joshua Colp * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c 2016-07-07 10:55 +0000 [fb96492ec4] Scott Griepentrog * PJSIP: provide valid tcp nodelay option for reuse When using TCP transport with chan_pjsip, the TCP_NODELAY option value was allocated on the stack, then passed as a pointer to the tcp transport configuration structure, and later re-used on subsequently created sockets when it was no longer valid. This patch changes the allocation to be a static. ASTERISK-26180 #close Reported by: Scott Griepentrog Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0 2016-07-06 09:29 +0000 [1c949eea6c] Alexei Gradinari * res_pjsip: Added "subscribe_context" to endpoint If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514 2016-07-04 05:58 +0000 [32cb981d04] Alexander Traud * BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is using AS_HELP_STRING everywhere else already. ASTERISK-26046 Change-Id: I8299faf504ceaeee3e39930c59293809e116c631 2016-06-22 17:26 +0000 [9f2c007254] Richard Mudgett * res_pjsip_session.c: Don't send extra BYE if SDP invalid. When an answer SDP is invalid we were disconnecting the outgoing call and sending two BYE requests. The first BYE was sent by PJPROJECT because of the invalid SDP answer. The second BYE was sent by Asterisk because it thought the canceled call was the result of the RFC5407 section 3.1.2 race condition. * Made not send the BYE on a canceled session if the SDP negotiation is incomplete because PJPROJECT has already sent a BYE for the failed negotiation. ASTERISK-25772 #close Reported by: Dmitriy Serov Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836 2016-06-27 17:19 +0000 [08d3b9a89e] Richard Mudgett * res_pjsip_session.c: End call on initial invalid SDP negotiation. When an incoming call defers SDP negotiation and then sends us an invalid SDP in the ACK, we need to send a BYE to disconnect the call. In this case SDP negotiation has failed and we don't have valid media streams negotiated. ASTERISK-25772 Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8 2016-06-23 15:13 +0000 [e6e12c752c] Richard Mudgett * res_pjsip.c: Register PJMEDIA error code decoder. Registering the PJMEDIA error codes allows errors found when parsing an incoming SDP to be easier to figure out. "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" is much easier to understand than "Unknown error 220030". ASTERISK-25772 Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0 2016-06-27 16:56 +0000 [5d2fc6bab7] Richard Mudgett * res_pjsip_session.c: Remove unused parameter from handle_incoming(). Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa 2016-06-22 18:02 +0000 [656ed73ac6] Richard Mudgett * res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047 2016-06-30 15:17 +0000 [4f7b859726] Richard Mudgett * features: Fix channel datastore access. Found as a result of the testsuite tests/callparking test crashing. Several calls to ast_get_chan_featuremap_config() and ast_get_chan_features_xfer_config() did not lock the channel before calling so the channel's datastore list was accessed without the lock's protection. Apparently another thread deleted a datastore on the channel's list while the crashing thread was walking the list. Crash at 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. * Add missing channel locks to calls that were not already protected as the doxygen for those calls indicates. Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1 2016-06-30 08:25 +0000 [5ad7e1c09a] gtjoseph * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK from getting set when using an external pjproject. ASTERISK-26099 #close Reported-by: Ross Beer Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae 2016-06-29 15:31 +0000 [dab2a6b689] Matt Jordan * hep.conf.sample: Default 'enabled' to 'no' Following the principle of least surprise, we should not be sending massive numbers of PJSIP and RTCP HEP packets out into the ether to some only-slightly-random IP address. Having 'enabled' set to 'no' in the sample configuration file should prevent this from happening for those who run 'make samples'. ASTERISK-26159 #close Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1 2016-06-29 15:09 +0000 [9129ac8e73] Matt Jordan * pjproject/patches/config_site: Increase the max number of ICE candidates When negotiating ICE candidates with WebRTC capable endpoints, many networks will result in a browser offering ICE candidates that exceeds the default number of max candidates, 16. This patch bumps the max candidates to 32, with the max checks at twice the number of candidates. In practice, this has shown to be sufficient for browser/WebRTC negotiation. Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5 2016-06-28 09:00 +0000 [4045e6d8ba] gtjoseph * codecs: Fix ABI incompatibility created by adding format_name to ast_codec Adding format_name even to the end of ast_codec caused issued with binary codec modules because the pointer would be garbage in asterisk when they registered. So, the ast_codec structure was reverted and an internal_ast_codec structure was created just for use in codec.c. A new internal-only API was also added (__ast_codec_register_with_format) so that codec_builtin could register codecs with the format_name in a separate parameter rather than in the ast_codec structure. ASTERISK-26144 #close Reported-by: Alexei Gradinari Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba 2016-06-28 08:22 +0000 [651290a809] gtjoseph * BuildSystem: Fix a few issues hightlighted by gcc 6.x gcc 6.1.1 caught a few more issues. Made sure the unit tests still pass for the func_env and stdtime issues. ASTERISK-26157 #close Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e 2016-06-28 10:33 +0000 [83f2c2573b] Matt Jordan * configs/basic-pbx/modules.conf: Remove 'bad' modules This patch removes the following modules: - pbx_functions: It never existed. - res_pjsip_log_forwarder: It no longer exists. - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs aren't going to be installing HOMER - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't loaded, and we aren't configured to make use of the module Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5 2016-06-22 11:19 +0000 [75818b4084] Joshua Colp * siren: Add format attribute modules for Siren7 and Siren14. This change removes hardcoded SDP parsing and generation for Siren7 and Siren14 from chan_sip and moves it to format attribute modules so it can also be used by chan_pjsip. With this the fmtp lines for both are added with the bitrate information. ASTERISK-26021 Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037 2016-06-23 04:33 +0000 [6e87bf746a] Alexander Traud * BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf. Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C but requires ANSI C anyway. ASTERISK-26046 Change-Id: I914c014385e1862102d90fe7650621def78db02e 2016-06-22 15:04 +0000 [8c7017f76e] Corey Farrell * res_fax: Fix reference leak in fax_v21_session_new. fax_v21_session_new created a session details object but only released the allocation reference during error conditions. fax_session_new adds it's own reference to details if needed so the caller is always responsible for cleaning it's own reference. ASTERISK-26141 #close Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88 2016-06-22 14:25 +0000 [6fa3ed0679] Alexei Gradinari * res_pjsip: improve realtime performance #2 The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df 2016-06-22 13:41 +0000 [d293ead077] gtjoseph * res_rtp_asterisk: Fix a self-comparison identified by gcc 6 gcc 6 caught a previously unidentified self-comparison in ice_candidate_cmp. Fixed it and re-ordered the predicates for better short-circuiting. ASTERISK-26140 #close Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7 2016-06-22 10:37 +0000 [c7309a5254] gtjoseph * chan_unistim: Fix memcpy in get_to_address A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) was using a pointer to a pointer as the destination of a memcpy and a '&' instead of '*' in the sizeof. ASTERISK-26138 #close Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708 2016-06-20 13:21 +0000 [b6bd97eea2] Mark Michelson * Fix Alembic upgrades. A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. In addition, the head of the alembic branch referred to a non-existent revision. This has been fixed by referring to the proper revision. This patch fixes another realtime problem as well. Our Alembic scripts store booleans as yes or no values. However, Sorcery tries to insert "true" or "false" instead. This patch introduces a new boolean type that translates to "yes" or "no" instead. ASTERISK-26128 #close Change-Id: I51574736a881189de695a824883a18d66a52dcef 2016-06-22 10:51 +0000 [3b4f5d1345] gtjoseph * test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO Since the file was missing the depends on pjproject, it wasn't picking up the pjproject related include path. If there was no system installed pjproject and pjproject-bundled was used, a compile would fail because pjsip.h wasn't found. ASTERISK-26139 #close Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757 2016-06-22 10:55 +0000 [5f23aacda4] Alexander Traud * BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf. Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not support the platform SVR2 from the year 1987 anymore. ASTERISK-26046 Change-Id: I28161b037feb2d29ab46ed20e785928460226c22 2016-06-21 06:52 +0000 [804005d251] Torrey Searle * res_rtp_asterisk: fix memory leak in dtls ensure that cert bios get freed after creating the fingerprint ASTERISK-26129 #close Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451 2016-06-21 17:42 +0000 [f572b26495] Richard Mudgett * res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c 2016-06-12 11:19 +0000 [b57cd01404] gtjoseph * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription Occasionally under load we'll attempt to send a final NOTIFY on a subscription that's already been terminated and a SEGV will occur down in pjproject's evsub_destroy function. This is a result of a race condition between all the paths that can generate a notify and/or destroy the underlying pjproject evsub object: * The client can send a SUBSCRIBE with Expires: 0. * The client can send a SUBSCRIBE/refresh. * The subscription timer can expire. * An extension state can change. * An MWI event can be generated. * The pjproject transaction timer (timer_b) can expire. Normally when our pubsub_on_evsub_state is called with a terminate, we push a task to the serializer and return at which point the dialog is unlocked. This is usually not a problem because the task runs immediately and locks the dialog again. When the system is heavily loaded though, there may be a delay between the unlock and relock during which another event may occur such as the subscription timer or timer_b expiring, an extension state change, etc. These may also cause a terminate to be processed and if so, we could cause pjproject to try to destroy the evsub structure twice. There's no way for us to tell that the evsub was already destroyed and the evsub's group lock can't tolerate this and SEGVs. The remedy is twofold. * A patch has been submitted to Teluu and added to the bundled pjproject which adds add/decrement operations on evsub's group lock. * In res_pjsip_pubsub: * configure.ac and pjproject-bundled's configure.m4 were updated to check for the new evsub group lock APIs. * We now add a reference to the evsub group lock when we create the subscription and remove the reference when we clean up the subscription. This prevents evsub from being destroyed before we're done with it. * A state has been added to the subscription tree structure so termination progress can be tracked through the asyncronous tasks. * The pubsub_on_evsub_state callback has been split so it's not doing double duty. It now only handles the final cleanup of the subscription tree. pubsub_on_rx_refresh now handles both client refreshes and client terminates. It was always being called for both anyway. * The serialized_on_server_timeout task was removed since serialized_pubsub_on_rx_refresh was almost identical. * Missing state checks and ao2_cleanups were added. * Some debug levels were adjusted to make seeing only off-nominal things at level 1 and nominal or progress things at level 2+. ASTERISK-26099 #close Reported-by: Ross Beer. Change-Id: I779d11802cf672a51392e62a74a1216596075ba1 2016-06-21 07:05 +0000 [6eb0354f2d] Alexander Traud * res_rtp_asterisk: Use latest DTLS version available by underlying platform. Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based cipher-suites. ASTERISK-26130 #close Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0 2016-06-21 10:53 +0000 [596d0b0bc3] Scott Griepentrog * PJSIP: provide transport type with received messages The receipt of a SIP MESSAGE may occur over any transport including TCP and TLS. When the message is received, the original URI is added to the message in the field PJSIP_RECVADDR, but this is insufficient to ensure a reply message can reach the originating endpoint. This patch adds the PJSIP_TRANSPORT field populated with the transport type. ASTERISK-26132 #close Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e 2016-06-21 08:01 +0000 [9e222efbf2] Alexander Traud * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. Some configure scripts used both AC_HELP_STRING and its replacement AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were changed to AS_HELP_STRING. ASTERISK-26046 Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f 2016-06-20 10:29 +0000 [e94aae00a7] Joshua Colp * res_pjsip_session: Handle race condition at shutdown with timer. When shutting down res_pjsip_session will get unloaded before res_pjsip. The act of unloading unregisters all the PJSIP services and sets their module IDs to -1. In some cases it is possible for a timer to occur after this happens which calls into res_pjsip_session. The res_pjsip_session module can then try to get the session from the INVITE session using the module ID. Since the module ID is now -1 this fails. This change stores a copy of the module ID and uses it for the timer callback scenario. If the module ID is -1 the callback immediately returns but if the module ID is valid then it continues as normal. This works as the original ID of the module is guaranteed to still be valid when used with the INVITE session. ASTERISK-26127 #close Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573 2016-06-20 12:13 +0000 [0a30008224] Richard Mudgett * app_voicemail.c: Fix IMAP compile error. Fix compile error introduced by the patch for ASTERISK-26045 Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3 2016-06-17 13:51 +0000 [820ed3d4b3] Alexei Gradinari * fix: memory leaks, resource leaks, out of bounds and bugs ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c 2016-06-13 17:40 +0000 [11caa10cf5] Mark Michelson * ARI: Ensure announcer channels are destroyed. Announcer channels were not being destroyed because the stasis_app_control structure that referenced them was not being destroyed. The control structure was not being destroyed because it was not being unlinked from its container. It was not being unlinked from its container because the after bridge callback for the announcer channel was not being run. The after bridge callback was not being run because the after bridge datastore was not being removed from the channel on destruction. The channel was not being destroyed because the hangup that used to destroy the channel was now only reducing the reference count to one. The reference count of the channel was only being reduced to one because the stasis_app_control structure was holding the final reference... The control structure used to not keep a reference to the channel, so that loop described above did not happen. The solution is to manually remove the control structure from its container when the playback on a bridge is complete. ASTERISK-26083 #close Reported by Joshua Colp Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4 2016-06-20 08:05 +0000 [f72ffc1ff9] Alexander Traud * http: leverage 'bindaddr' for TLS in http.conf The internal HTTP/WebSocket server supports both TCP and TLS, which can be activated separately via the file http.conf. The source code intends to re-use the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified explicitly. This did not work because of a typo. This change resolves this typo. ASTERISK-26126 #close Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f 2016-05-18 17:37 +0000 [3c80f84cd0] Richard Mudgett * res_pjsip_transport_management.c: Misc cleanups to survive shutdown. * In unload_module(), reordered destroying things to minimize the window that the global transports container could be used by other threads on shutdown. When shutting down you need to stop things in the opposite order of creation. * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to eliminate the crash potential by other threads using the container on shutdown. * Made struct monitored_transport.sip_received not use ast_atomic_fetchadd_int() since it is used as a boolean value that is only set TRUE. It was previously incremented for every received SIP message and could theoretically overflow. * In monitored_transport_state_callback(), allocated the monitored transport object without a lock since the lock was unused. * In keepalive_global_loaded(), removed releasing the transports container if the keepalive_thread could not be started. I set it up to be tried again if the user reloads the configuration. Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff 2016-01-05 19:08 +0000 [7c59f2126f] Richard Mudgett * res_pjsip.c: Add check that timer actually got scheduled. Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1 2016-06-13 13:33 +0000 [51cc5c31c4] Richard Mudgett * res_rtp_multicast.c: Fix warning message typo. Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3 2016-02-11 18:15 +0000 [3d0632a9c2] Richard Mudgett * res_pjsip_session.c: Reorganize ast_sip_session_terminate(). Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b 2016-06-08 06:15 +0000 [ac683f13c9] Alexander Traud * core: Not the configured but granted number of possible file descriptors. With CLI "core show settings", simply the parameter maxfiles of the file asterisk.conf was shown. If that parameter was not set, nothing was displayed although the environment might have set a default number itself. Or if maxfiles were not granted (completely), still maxfiles was shown. Now, the maximum number of possible file descriptors in the environment is shown. ASTERISK-26097 Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b 2016-06-10 10:39 +0000 [4eb8cf2684] Joshua Colp * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. This reverts commit 5bfef2a8b4674382f959b21a3b8e14cf1d942bab as it caused fax test failures. ASTERISK-25629 Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4 2016-06-08 06:05 +0000 [0bf1a53db3] Alexander Traud * astfd: With RLIMIT_NOFILE only the current value is sensible. With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max and current max of possible file descriptors were shown. Both show the same value always. Not to confuse users, just the current maximum is shown now. ASTERISK-26097 Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa 2016-06-07 18:45 +0000 [d338343dac] Joshua Colp * cel: Ensure only one dial status per channel exists. CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe 2016-06-01 13:48 +0000 [1fd3a7849e] Mark Michelson * ARI: Ensure proper channel state on operations. ARI was recently outfitted with operations to create and dial channels. This leads to the ability to try funny stuff. You could create a channel and then immediately try to play back media on it. You could create a channel, dial it, and while it is ringing attempt to make it continue in the dialplan. This commit attempts to fix this by adding a channel state check to operations that should not be able to operate on outbound channels that have not yet answered. If a channel is in an invalid state, we will send a 412 response. ASTERISK-26047 #close Reported by Mark Michelson Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8 2016-06-08 11:27 +0000 [10019dc70c] Mark Michelson * test_http_media_cache: Fix failing test. The retrieve_cache_control_directives test has been failing occasionally in Jenkins. The apparent failure occurs when attempting to validate the expiration of the retrieved file. After reproducing, the problem was pretty clear. At the beginning of the test, the current time is retrieved. The seconds value of this timestamp is X. When the file is retrieved, res_http_media_cache calculates the expiration and in doing so retrieves the current time. In most cases, since the test executes quickly, it will also retrieve a timestamp with X seconds. However, if the test starts very near to when the timestamp seconds are set to increment, res_http_media_cache may retrieve a timestamp with X+1 seconds instead. The test attempted to account for this by allowing a tolerance of 1 second when validating the expiration. However, the problem was that the comparisons being used in the validation used > and < operations. This meant that values that fell within the tolerance (because they equaled the upper bound of the tolerance) would fail. The solution is to use >= and <= operators in the expiration validation. However, I estimated that while the one second tolerance should be fine on most machines, it would still be possible on a very slow machine to end up falling outside the one second tolerance. So I have also relaxed the tolerance of expiration validation to be three seconds instead. The final change here is to add a debug message when validating expiration so that we can see what values are being compared. ASTERISK-25959 #close Reported by Joshua Colp Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247 2016-06-03 01:20 +0000 [56bdf048d2] Timo Teräs * Add support for OGG/Speex file format ASTERISK-18995 #close Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a 2016-06-09 10:33 +0000 [f0855358a6] gtjoseph * cdr.c: Remove assert in base_process_dial_end Scenario: Caller blonde transfer Bob calls Charlie who answers. Bob puts Charlie on hold and calls Alice. Before Alice answers, Bob transfers Charlie to Alice. Charlie's channel triggers an assert because he gets an "ANSWERED" event even though he never dialed anything. With recent changes to dial events, this is now a valid scenario so the assert needed to be removed. ASTERISK-26103 #close Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2 2016-06-09 10:37 +0000 [cdb7edbe7b] Mark Michelson * chan_pjsip: Lock channel when checking for RTP changes. bridge_native_rtp can call into an RTP-capable channel driver in order for the driver to update information about who the channel is communicating with. For SIP channel drivers, this means deactivating RTCP and sending a reinvite so that the endpoints can communicate directly. bridge_native_rtp does the right thing and has the channel locked when calling into the channel driver. chan_pjsip can't alter session properties in this thread, though. chan_pjsip queues a task on the session serializer in order to update properties there. The problem is that this queued task was not locking the channel. This meant that the queued task could attempt to deactivate RTCP at the same time that the channel thread was attempting to process an incoming RTCP packet. This could lead to a crash. This patch fixes the issue by locking the channel in the queued task when altering RTP properties. ASTERISK-26092 #close Reported by Niklas Larsson Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159 2016-06-03 22:44 +0000 [04ec9c745e] Richard Mudgett * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. This patch fixes a race condition processing received REGISTER requests and their retransmissions caused by REGISTER requests being processed by two threads. The "sip_transaction Unable to register REGISTER transaction (key exists)" message is a notable symptom of this issue. This issue was more likely to happen before the pjsip/distributor serializers were created. Instead of steps one and two below placing the REGISTER messages into the same pjsip/distributor they were placed in random pjsip/default serializers. 1) REGISTER requests come in and get placed on the pjsip/distributor serializer. 2) Before the first request is processed a retransmission comes in and is placed on the same pjsip/distributor serializer. 3) The first request goes up the pjsip stack and is then shunted off to the pjsip/aor/ serializer. 4) Before the first request is completed processing in the pjsip/aor/ serializer, the second request goes up the pjsip stack and is also shunted off to the pjsip/aor/ serializer. 5) The first request completes processing and sends out its response. 6) The second request completes processing and tries to send out its response but pjlib complains that the REGISTER transaction key already exists. 7) Sadness ensues. * The race is eliminated by removing the pjsip/aor/ serializer and continuing the processing in the pjsip/distributor serializer. Now any retransmissions queued in the pjsip/distributor serializer will be processed after the first message is completely processed. ASTERISK-26088 #close Reported by: Richard Mudgett Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a 2016-06-03 11:35 +0000 [dcfef53ee2] Richard Mudgett * stasis: Add setting subscription congestion levels. Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244 2016-06-02 18:19 +0000 [4879cd875c] Richard Mudgett * sorcery: Add setting object type congestion levels. Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 2016-06-02 16:08 +0000 [2cd67d5b07] Richard Mudgett * taskprocessors: Implement high/low water mark alerts. When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa 2016-05-27 17:31 +0000 [c966a035e0] Richard Mudgett * res_pjsip_session: Use distributor serializer for incoming calls. We must continue using the serializer that the original INVITE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. Outgoing call legs create the pjsip/outsess/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc 2016-05-27 16:28 +0000 [5b7b16a87f] Richard Mudgett * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be 2016-05-27 12:50 +0000 [c2ae49249c] Richard Mudgett * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0 2016-05-26 17:35 +0000 [2ff26e9746] Richard Mudgett * pjsip_distributor.c: Consistently pick a serializer for messages. Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407 2016-06-02 12:51 +0000 [df2791da8f] Richard Mudgett * pjsip_distributor.c: Ignore messages until fully booted. We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 2016-06-09 09:20 +0000 [d21a77b325] gtjoseph * build: Fix ast_sockaddr initialization to be more portable A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4 2016-06-03 00:59 +0000 [72d190eb69] Timo Teräs * Detect and use proper libraries for musl toolchains Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb 2016-06-03 00:57 +0000 [39b69ab537] Timo Teräs * Fixes to include signal.h POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc 2016-06-08 12:26 +0000 [7f5ca67e5f] Matt Jordan * res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea 2016-06-08 02:11 +0000 [784c18128b] Alexander Traud * chan_sip: No rtpmap for static RTP payload IDs in SDP. This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compactheaders=yes via the file sip.conf. ASTERISK-25578 #close Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044 2016-06-02 12:04 +0000 [31a5c28339] Joshua Colp * res_odbc: Implement a connection pool. Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff 2016-05-31 09:10 +0000 [80ff7912a1] Vasil Kolev * chan_sip: bigger buffers for headers, better failure mode Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3 2016-06-06 11:13 +0000 [60caebc738] Örn Arnarson * apps/app_voicemail.c and main/say.c: Add support for Icelandic language Icelandic has some weird grammar rules when dealing with dates and numbers. There are different genders used depending on which number you're dealing with, and only a handful of numbers do change depending on the gender. There is also an implied gender in several cases. This patch was originally written for asterisk 1.6, and has been in use for several years without crashes. I cleaned it up a bit and rewrote what was necessary for Asterisk 13. The functions were copied from other similar languages and modified where appropriate. If i recall correctly, the German and Danish functions were used as a base. ASTERISK-26087 Reported by: Örn Arnarson Tested by: Örn Arnarson Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665 2016-06-07 05:45 +0000 [52120204c9] Alexander Traud * res_srtp: Instead of libSRTP use OpenSSL as random source. Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. Therefore, the symbol RAND_bytes is used instead of crypto_get_random. ASTERISK-24436 #close Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96 2016-06-07 02:16 +0000 [da943ec5c0] Alexander Traud * BuildSystem: Avoid 'ar cru' and use 'ar cr' instead. In several internal library projects, the files are archived with the help of 'ar cr'. Only the projects editline and the Objective Open H.323 stack implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier ignored since `D' is the default (see `U')". For consistency and to avoid this message all projects use 'ar cr' now. ASTERISK-26091 #close Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40 2016-06-01 16:57 +0000 [dca052e531] Richard Mudgett * chan_rtp.c: Simplify options to UnicastRTP channel creation. Change the awkward and not as flexible UnicastRTP options format From: Dial(UnicastRTP/127.0.0.1[/[][/[]]]) To: Dial(UnicastRTP/127.0.0.1[/[]]) Where can be standard Asterisk flag options: c() - Specify which codec/format to use such as 'ulaw'. e() - Specify which RTP engine to use such as 'asterisk'. More option flags can be easily added later such as the codec's RTP payload type to use when the codec does not have a static payload type defined. Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9 2016-05-02 05:57 +0000 [5bfef2a8b4] Jaco Kroon * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ASTERISK-25629 #close Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006 2016-05-25 10:34 +0000 [3e8d523d88] Alexei Gradinari * core/dial: New channel variable FORWARDERNAME Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 2016-05-27 14:49 +0000 [a2f820e8dc] gtjoseph * ari/resource_channels: Add 'formats' to channel create/originate If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b 2016-06-03 01:33 +0000 [538c6415c6] Timo Teräs * chan_sip: Support auth username for callbackextension feature ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239 2016-06-03 00:39 +0000 [797695c5cc] Timo Teräs * Make use of GLOB_BRACE and GLOB_NOMAGIC optional These flags are non-portable GNU extensions. Make their use optional. This fixes complication error on e.g. musl c-library based systems. Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3 2016-06-02 14:57 +0000 [3c1fec8099] Timo Teräs * Fix res_search usage Resolver state is not part of res_search API. This fixes compilation error: dns.c:261:8: error: too many arguments to function 'res_search' ret = res_search(&dns_state, Change-Id: Ia600a58557040df83f744da3dde23225293845a5 2016-06-02 14:53 +0000 [9c1d95e873] Timo Teräs * Fix #include poll.h and sys/cdefs.h POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be 2016-05-25 08:45 +0000 [8a5c2e736c] Niklas Larsson * core/manager: Add uptime field to FullyBooted Add Uptime and LastReload to event FullyBooted. ASTERISK-26058 #close Reported by: Niklas Larsson Change-Id: I909b330801c0990d78df9b272ab0adc95aecb15e 2016-06-02 04:59 +0000 [4505a59dc9] Joshua Colp * alembic: Fix migration. The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting to use UniqueConstraint and failing. It was not imported and after importing it also continued to fail. I've changed the script to use the explicit name of the constraint instead. Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9 2016-06-01 13:57 +0000 [40d19f2e55] Richard Mudgett * logging,cdr,cel: Fix stringfield memory leak. The stringfields refactor to allow adding stringfields to the end of a structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some incomplete cleanup code by some stringfield users. The most noticeable leaker is the logging system where there is a leak for every log message generated. ASTERISK-26078 #close Reported by: Etienne Lessard Patches: jira_asterisk_26078_v13.patch (license #5621) patch uploaded by Richard Mudgett Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782 2016-05-31 13:02 +0000 [aec7916595] Richard Mudgett * pjsip_distributor.c: Use correct rdata info access method (Part 2). The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799 2016-05-09 15:00 +0000 [205a31f86c] Mark Michelson * Expand the scope of Dial Events Dial events up to this point have come in two flavors * A Dial event with no status to indicate that dialing has begun * A Dial event with a status to indicate that dialing has ended With this change, Dial events have been expanded to also give intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". This is especially useful for ARI dialing, as it gives the application writer the opportunity to place a channel into an early bridge when early media is detected. AMI handles these in-progress dial events by sending a new event called "DialState" that simply indicates that dial state has changed but has not ended. ARI never distinguished between DialBegin and DialEnd, so no change was made to the event itself. Another change here relates to dial forwards. A forward-related event was previously only sent when a channel was successfully able to forward a call to a new channel. With this set of changes, if forwarding is blocked, we send a Dial event with a forwarding destination but no forwarding channel, since we were prevented from creating one. This is again useful for ARI since application writers can now handle call forward attempts from within their own application. ASTERISK-25925 #close Reported by Mark Michelson Change-Id: I42cbec7730d84640a434d143a0d172a740995543 2016-05-30 19:27 +0000 [8a6a14590d] gtjoseph * res_pjsip_mwi_body_generator: Re-order the body items Re-ordered the body items so Message-Account is second. Messages-Waiting: no Message-Account: sip:1571@:5060 Voice-Message: 0/0 (0/0) ASTERISK-26065 #close Reported-by: Ross Beer Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3 2016-05-30 10:58 +0000 [7fa5766752] gtjoseph * pjproject_bundled: Move to pjproject 2.5 Although all the patches we had against 2.4.5 were applied by Teluu, a new bug was introduced preventing re-use of tcp and tls transports This patch removes all the previous patches against 2.4.5, updates the version to 2.5, and adds a new patch to correct the transport re-use problem. Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068 2016-05-27 12:25 +0000 [b56f611856] Rusty Newton * res_pjsip: Add clarifying documentation to PJSIP_HEADER help text Added notes about when you can read or write headers. Specifically about being able to read on the inbound channel and write on an outbound channel. ASTERISK-26063 #close Reported by: Private Name Tested by: Rusty Newton Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5 2016-05-26 15:14 +0000 [bb0f4a6310] Mark Michelson * multicast RTP: Add dialing options This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219 2016-05-09 14:48 +0000 [88d997913f] Mark Michelson * ARI: Re-implement the ARI dial command, allowing for early bridging. ARI dial had been implemented using the Dial API. This made great sense when dialing was 100% separate from bridging. However, if a channel were to be added to a bridge during the dial attempt, there would be a conflict between the dialing thread and the bridging thread. Each would be attempting to read frames from the dialed channel and act on them. The initial attempt to make the two play nice was to have the Dial API suspend the channel in the bridge and stay in charge of the channel until the dial was complete. The problem with this was that it was riddled with potential race conditions. It also was not well-suited for the case where the channel changed which bridge it was in during the dial. This new approach removes the use of the Dial API altogether. Instead, the channel we are dialing is placed into an invisible ARI dialing bridge. The bridge channel thread handles incoming frames from the channel. If the channel is added to a real bridge, it is departed from the invisible bridge and then added to the real bridge. Similarly, if the channel is removed from the real bridge, it is automatically added back to the invisible bridge if the dial attempt is still active. This approach keeps the threading simple by always having the channel being handled by bridge channel threads. ASTERISK-25925 Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb 2016-05-19 14:56 +0000 [31f17abe44] Alexei Gradinari * res_pjsip: add "via_addr", "via_port", "call_id" to contact As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576 2016-05-24 16:56 +0000 [574c9e77eb] Alexei Gradinari * res_pjsip: chatty verbose messages There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7 2016-05-20 13:56 +0000 [b3142e99e4] Alexei Gradinari * app_voicemail: fix bugs, imap mm_status log change to debug Fixed some bugs: - create dirpath when save downloading message from IMAP storage. - create IMAP folder if not exists when saving to IMAP storage - check if file successfully opened before write to it - some IMAP checks - remove non-standard flag 'Unseen' etc Change to debug IMAP mm_status log instead of verbose. Remove unused X-Asterisk-VM-Caller-channel message header for security reason. The clients should not know name of peer/endpoint. ASTERISK-26045 #close Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b 2016-05-25 18:30 +0000 [7d44d12816] Richard Mudgett * pjsip_distributor.c: Use correct rdata info access method. The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2 2016-05-03 11:11 +0000 [1d60bfcdf1] Tzafrir Cohen * followme: allow disabling callee prompt Add the option 'enable_callee_prompt' to followme.conf. Enabled by default. If disabled, a callee is not prompted to accept or reject the forwarded call. ASTERISK-26064 #close Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5 Signed-off-by: Tzafrir Cohen 2016-02-12 09:59 +0000 [80ff2c2540] Corey Farrell * threadpool: Fix potential data race. worker_start checked for ZOMBIE status without holding a lock. All other read/write of worker status are performed with a lock, so this check should do the same. ASTERISK-25777 #close Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781 2016-05-24 05:28 +0000 [070eab6ed2] Joshua Colp * res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying. Recent changes to res_pjsip_outbound_publish have introduced a race condition at shutdown where an outbound publish may be shutdown twice. In this case the first succeeds as a result of the unpublish. In the second invocation since it's been unpublished a task is queued to just destroy the client. This task holds no ref to the publish and as a result the publish may be destroyed before the task is run, causing a crash. This explicit destruction task now holds a reference to the publish to ensure it remains valid. ASTERISK-26053 #close Change-Id: I10789b98add3e50292ee3b33a55a1d9061cec94b 2016-05-09 14:27 +0000 [f6c33771f6] Mark Michelson * Bridging: introduce "invisible" bridges. Invisible bridges function the same as normal bridges, but they have the following restrictions: * They never show up in CLI, AMI, or ARI queries. * They do not have Stasis messages published about them. Invisible bridges' main use is for when use of the bridging system is desired, but the bridge should not be known to users of the Asterisk system. ASTERISK-25925 Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670 2016-05-22 11:03 +0000 [85d0272e76] Joshua Colp * res_pjsip: Only check transaction on transaction state events. The send request callback function currently assumes that it will only ever be called on transaction state changes. This is not always true. If our own timer callback occurs we will call the callback with a timer event instead of a transaction state change event. In this case the transaction on the event is invalid and accessing it will result in a crash. ASTERISK-26049 #close Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc 2016-05-21 05:42 +0000 [31897d2d99] Jesper (License 5518) * func_curl: Don't trim response text on non-ASCII characters The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of a signed comparison. ASTERISK-25669 #close Reported by: Jesper patches: strings.curl.trim.patch submitted by Jesper (License 5518) Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a 2016-05-20 19:03 +0000 [2a77af9ed0] Richard Mudgett * chan_rtp.c: Cleanup ast_request() parameter parsing. * Fixed NULL crash potential if parameters are missing. * Reordered some operations so further diagnostic messages can be more helpful. Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70 2016-05-20 16:59 +0000 [ade5275a3e] Richard Mudgett * parking.h: Update ast_parking_park_call() doxygen to reality. ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260 2016-05-12 15:18 +0000 [c378b00a83] Alexei Gradinari * func_odbc: single database connection should be optional func_odbc was changed in Asterisk 13.9.0 to make func_odbc use a single database connection per DSN because of reported bug ASTERISK-25938 with MySQL/MariaDB LAST_INSERT_ID(). This is drawback in performance when func_odbc is used very often in dialplan. Single database connection should be optional. ASTERISK-26010 Change-Id: I7091783a7150252de8eeb455115bd00514dfe843 2016-05-20 09:39 +0000 [1c02b19b79] Mark Michelson * res_pjsip: Match dialogs on responses better. When receiving an incoming response to a dialog-starting INVITE, we were not matching the response to the INVITE dialog. Since we had not recorded the to-tag to the dialog structure, the PJSIP-provided method to find the dialog did not match. Most of the time, this was not a problem, because there is a fall-back that makes the response get routed to the same serializer that the request was sent on. However, in cases where an asynchronous DNS lookup occurs in the PJSIP core, the thread that sends the INVITE is not actually a threadpool serializer thread. This means we are unable to record a serializer to handle the incoming response. Now, imagine what happens when an INVITE is sent on a non-serialized thread, and an error response (such as a 486) arrives. The 486 ends up getting put on some random threadpool thread. Eventually, a hangup task gets queued on the INVITE dialog serializer. Since the 486 is being handled on a different thread, the hangup task can execute at the same time that the 486 is being handled. The hangup task assumes that it is the sole owner of the INVITE session and channel, so it ends up potentially freeing the channel and NULLing the session's channel pointer. The thread handling the 486 can crash as a result. This change has the incoming response match the INVITE transaction, and then get the dialog from that transaction. It's the same method we had been using for matching incoming CANCEL requests. By doing this, we get the INVITE dialog and can ensure that the 486 response ends up being handled by the same thread as the hangup, ensuring that the hangup runs after the 486 has been completely handled. ASTERISK-25941 #close Reported by Javier Riveros Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0 2016-05-18 06:19 +0000 [e773e3a9bb] Matt Jordan * ARI: Add the ability to download the media associated with a stored recording This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181 2016-05-19 11:41 +0000 [40cb032009] Joshua Colp * res_sorcery_astdb: Filter fields to only the registered ones. This change introduces the same filtering that is done in res_sorcery_realtime to the res_sorcery_astdb module. This allows persisted sorcery objects that may contain unknown fields to still be read in from the AstDB and used. This is particularly useful when switching between different versions of Asterisk that may have introduced additional fields. ASTERISK-26014 #close Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2 2016-05-09 21:40 +0000 [9766a12b4c] snuffy * res_pjsip_empty_info: Respond to empty SIP INFO packets Some SBCs require responses to empty SIP INFO packets after establishing call via INVITE, if not responded to they may drop your call after unspecified timeout of X minutes. They are identified by having no Content-Type, check for this and respond with 200 - OK message. ASTERISK-24986 #close Reported-by: Ilya Trikoz, Federico Santulli Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0 2016-05-18 10:58 +0000 [111c4b0324] Tzafrir Cohen * Makefile: remove OSARCH check for init install There are more specific checks for the platform. Specifically this allows installing OS/X init scripts. ASTERISK-26038 #close Change-Id: If08933621145b10362a0cfe73c079301d9c13f50 Signed-off-by: Tzafrir Cohen 2016-05-10 11:28 +0000 [d4b77dad1b] Joshua Colp * res_pjsip_exten_state: Use the extension for publishing to. This change uses the newly added multi-user support for outbound publish to publish to the specific user that an extension state change is for. This also extends the res_pjsip_outbound_publish support to include the user specific From and To URI information in the outbound publishing of extension state. Since the URI is used when constructing the body it is important to ensure that the correct local and remote URIs are used. Finally the max string growths for the dialog-info+xml body generator has been increased as through testing it has proven to be too conservative. ASTERISK-25965 Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1 2016-05-03 16:07 +0000 [3905997bae] Kevin Harwell * res_pjsip_outbound_publish: Add multi-user support per configuration Added a new multi_user option that when specified allows a particular configuration to be used for multiple users. It does this by replacing the user portion of the server uri with a dynamically created one. Two new API calls have been added in order to make use of the new functionality: ast_sip_publish_user_send - Sends an outgoing publish message based on the given user. If state for the user already exists it uses that, otherwise it dynamically creates new outbound publishing state for the user at that time. ast_sip_publish_user_remove - Removes all outbound publish state objects associated with the user. This essentially stops outbound publishing for the user. ASTERISK-25965 #close Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc 2016-05-18 07:54 +0000 [6e5e84458f] gtjoseph * udptl: Don't eat sequence numbers until OK is received Scenario: Local fax -> Asterisk w/ firewall -> Provider -> Remote fax * Local fax starts rtp call to remote fax * Remote fax starts t38 call back to local fax. * Local fax sends t38 no-signal to Asterisk before sending an OK. * udptl processes the frame and increments the expected sequence number. * chan_sip drops the frame because the call isn't up so nothing goes out the external interface to open the port for incoming packets. * Local fax sends OK and Asterisk sends OK to the remote fax. * Remote fax sends t38 packets which are dropped by the firewall. * Local fax re-sends t38 no-signal with the same sequence number. * udptl drops the frame because it thinks it's a dup. * Still no outgoing packets to open the firewall. * t38 negotiation fails. The patch drops frames t38 received before udptl sequence processing when the call hasn't been answered yet. The second no-signal frame is then seen as new and is relayed out the external interface which opens the port and allows negotiation to continue. ASTERISK-26034 #close Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 2016-05-15 12:22 +0000 [52148d93f4] Matt Jordan * CHANGES: Update formatting of items * Provide consistent indenting of lines in bulleted paragraphs * Respect the 80 character column width * Group all like items together, e.g., all dialplan applications under "Applications", etc. * Use a single blank line to break up functionality changes within a larger section * Use two blanks lines to delineate larger sections Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647 2016-04-18 18:17 +0000 [03d88b5656] Matt Jordan * ARI: Add the ability to play multiple media URIs in a single operation Many ARI applications will want to play multiple media files in a row to a resource. The most common use case is when building long-ish IVR prompts made up of multiple, smaller sound files. Today, that requires building a small state machine, listening for each PlaybackFinished event, and triggering the next sound file to play. While not especially challenging, it is tedious work. Since requiring developers to write tedious code to do normal activities stinks, this patch adds the ability to play back a list of media files to a resource. Each of the 'play' operations on supported resources (channels and bridges) now accepts a comma delineated list of media URIs to play. A single Playback resource is created as a handle to the entire list. The operation of playing a list is identical to playing a single media URI, save that a new event, PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final media URI. When the entire list is finished being played, a PlaybackFinished event is raised. In order to help inform applications where they are in the list playback, the Playback resource now includes a new, optional attribute, 'next_media_uri', that contains the next URI in the list to be played. It's important to note the following: - If an offset is provided to the 'play' operations, it only applies to the first media URI, as it would be weird to skip n seconds forward in every media resource. - Operations that control the position of the media only affect the current media being played. For example, once a media resource in the list completes, a 'reverse' operation on a subsequent media resource will not start a previously completed media resource at the appropiate offset. - This patch does not add any new operations to control the list. Hopefully, user feedback and/or future patches would add that if people want it. ASTERISK-26022 #close Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f 2016-05-17 11:14 +0000 [5bd1bf2816] gtjoseph * chan_sip: Prevent extra Session-Expires headers from being added When chan_sip does a re-INVITE to refresh a session and authentication is required, the INVITE with the Authorization header containes a second Session-Expires header without the ";refersher=" parameter. This is causing some proxies to return a 400. Also, when Asterisk is the uas and the refresher, it is including the Session-Expires and Min-SE headers in OPTIONS messages which is not allowed per RFC4028. This patch (based on the reporter's) Checks to see if a Session-Expires header is already in the message before adding another one. It also checks that the method is INVITE or UPDATE. ASTERISK-26030 #close Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 2016-05-16 15:29 +0000 [ae81b55361] gtjoseph * res_pjsip_outbound_registration: Clean up state when registration is deleted Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23 2016-05-15 19:05 +0000 [8b5cee4a4f] gtjoseph * res_pjsip: Set TCP_NODELAY on TCP transports Although it's perfectly legal to place multiple SIP messages in the same packet, it can cause problems because the Linux default is to enable Path MTU Discovery which sets the Don't Fragment bit on the packets. If adding a second message to the packet causes the MTU to be exceeded, and the destination isn't equipped to send a FRAGMENTATION NEEDED response to a large packet, the packet will just be dropped. We can't specifically tell the stack to send only 1 message per packet, but we can turn on TCP_NODELAY when we create the transport. This will at least tell the stack to send packets as soon as possible. ASTERISK-26005 #close Reported-by: Ross Beer Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd 2016-05-14 07:24 +0000 [3522376512] Matt Jordan * logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96 2016-05-14 21:48 +0000 [a1803cb5f4] Matt Jordan * configs/samples/pjsip.conf.sample: Fix typo A ':' is not a valid token for starting a comment. Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad 2016-05-12 07:08 +0000 [d29c17834c] Matt Jordan * res/res_hep_pjsip: Fix reported local IP address when bound to 'any' When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3 2016-05-14 12:29 +0000 [14938184a3] Sean Bright * res_ari: Correct Location headers returned by some ARI resources The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c 2016-05-11 20:17 +0000 [e06a23681c] Matt Jordan * res_hep: Provide an option to pick the UUID type At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c 2016-05-13 11:46 +0000 [69a85a519f] Alexei Gradinari * res_pjsip: Endpoint IP Access Controls With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02 2016-05-12 14:36 +0000 [fd3f70598d] Mark Michelson * Use doubles instead of floats for conversions when comparing strings. In 13.9.0, there was an issue where PJSIP contacts added to an AOR would be deleted at seemingly random times. One reason this was happening was because of an operation to retrieve the contacts whose expiration time was less than or equal to the current time. When retrieving existing contacts, the contact's expiration time and the current time were converted from a string to a float, and those two floats were compared. On some systems, including mine, this conversion was horribly off. For instance, I could regularly see the string "1463079214" get converted into 1463079168.000000. When switching from using a float to using a double, the conversion was as expected. Why was the conversion to float off? My best guess is that the conversion to float was attempting to store the entire value in the 23 bit significand of the IEEE-754 floating point number. In particular, if you take only the 23 most significant bits of 1463079214, you get the messed up 1463079168 that we were seeing in the conversion. It likely was possible to get a more precise value by composing the number using an exponent, but the conversion did not work that way. With a double, you have a 52 bit significand, allowing the entire value to fit there, and thereby allowing an accurate conversion. ASTERISK-26007 #close Reported by Greg Siemon Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070 2016-05-12 09:13 +0000 [4f8cfa0220] gtjoseph * pjsip_distributor: Add missing newline to NOTICE There was a newline missing from the end of the "no matching endpoint" notice. Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181 2016-05-10 10:19 +0000 [d14d1ba826] Sebastian Damm * res_pjsip_outbound_registration: generate correct Contact URI for TLS There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2 2016-05-05 16:41 +0000 [9f996624b0] Alexei Gradinari * logger: Add PID to syslog messages. During refactoring of this support the addition of the PID to messages was removed. This change adds it back in. ASTERISK-25538 #close Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36 2016-05-11 14:07 +0000 [5236ffed97] Matt Jordan * configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER When running on a system that does not support or use AST_UNDEFINED_SANITIZER or AST_LEAK_SANITIZER, the configure script would incorrectly set those constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would cause menuselect to error out, complaining that a blank value is not a valid option. This patch corrects the issue by setting the value to 0 if the options that those constants enable/disable is not found. Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba 2016-05-10 08:17 +0000 [b5c471b339] Tzafrir Cohen * followme: delete the right recorded name file FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen 2016-05-10 03:10 +0000 [ec85ea3c21] Tzafrir Cohen * basic-cfg: asterisk.conf: don't set languages * No need to set language in a miniml configuration. 'en' will do just fine. * It would be useful to have an example of setting it to a different language. * Setting the documentation language explicitly is likewise not required. Setting it to a different value is not common. At least until there is a set of translated documentation. Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 Signed-off-by: Tzafrir Cohen 2016-05-10 03:08 +0000 [1b0a9bb2c4] Tzafrir Cohen * basic-cfg: asterisk.conf: debug level 5 spams Don't suggest users to use debug level 5, which spews (usually non-useful) debug information. Reduce the suggestion to (an arbitrarily-selected) level 2. Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 Signed-off-by: Tzafrir Cohen 2016-05-10 03:06 +0000 [d0ba3e8196] Tzafrir Cohen * basic-cfg: asterisk.conf: defaults of options Note the default of remmed-out options. To clarify that those values are not the defaults. Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 Signed-off-by: Tzafrir Cohen 2016-05-10 02:56 +0000 [f943a1fd84] Tzafrir Cohen * basic-cfg: asterisk.conf: remove [directories] A minimal configuration does not need to explicitly spell out the directories. The built-in defaults will do just fine. In many cases they are wrong. Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c Signed-off-by: Tzafrir Cohen 2016-05-05 11:37 +0000 [1e876d6915] Kevin Harwell * res_pjsip_authenticator_digest: Don't use source port in nonce verification From the issue reporter: "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of the timestamp, the source address, the source port, a server UUID that is calculated at startup, and the authentication realm. Rather than caching nonces that we create, we instead attempt to re-calculate the nonce when receiving an incoming request with authentication. We then compare the re-calculated nonce to the incoming nonce, and if they don't match, then authentication has failed early. The problem is that it is possible, especially when using TCP, to receive two requests from the same endpoint but have differing source ports for those requests. Asterisk itself commonly will use different source ports for outbound TCP requests." This patch removes the source port dependency when building the nonce. ASTERISK-25978 #close Change-Id: I871b5f4adce102df1c4988066283095ec509dffe 2016-05-07 14:39 +0000 [dfefbf8731] gtjoseph * config_transport: Tell pjproject to allow all SSL/TLS protocols The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. SSL is not allowed. So, even if you specify "sslv3" for a transport method, it's silently ignored and one of the TLS protocols is used. This was a new behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that we never caught. Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). This tells pjproject to set the socket protocol to match the method. ASTERISK-26004 #close Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078 2016-05-05 09:14 +0000 [d03e170ae7] Joshua Colp * res_pjsip_pubsub: Use common datastores container API. This migrates res_pjsip_pubsub over to using the newly introduce common datastores management API instead of using its own implementations for both subscriptions and publications. As well the extension state data now provides a generic datastores container instead of a subscription. This allows the dialog-info+xml body generator to work for both subscriptions and publications. ASTERISK-25999 #close Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1 2016-05-05 09:12 +0000 [94cd351ec4] Joshua Colp * datastore: Add common container based datastores API. This change introduces a common container based datastores management API. This has been done in a few places across the tree but this consolidates all of the logic into one place in a generic fashion. ASTERISK-25999 Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a 2016-05-04 02:40 +0000 [8923c9ac96] Jaco Kroon * app_confbridge: Add a regcontext option for confbridge bridge profiles. This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f 2016-05-08 20:19 +0000 [facce6f632] gtjoseph * pjproject_bundled: Check for python-dev and TEST_FRAMEWORK The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. The python bindings are now built only if TEST_FRAMEWORK is set and a python development package is installed. libresample was also disabled. ASTERISK-25993 #close Reported-by: Joshua Colp Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03 2016-05-06 11:54 +0000 [322c3b4262] Alexei Gradinari * res_pjsip: module load priority The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* and res_pjsip_registrar modules should load ASAP to avoid "No matching endpoint found" for legitimate endpoint. ASTERISK-25994 Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b 2016-05-05 15:16 +0000 [516f49f316] Alexei Gradinari * stasis_endpoints: Add new Status and Headers to ContactStatus ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. These additions should be also in stasis_endpoints to include in command "manager show event ContactStatus" Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a 2016-05-03 15:43 +0000 [64e058f75a] Kevin Harwell * res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches When reloading, or fetching realtime data, if the "apply" failed for any numerous reasons the current state object would not be maintained. This potentially resulted in publishes being stopped for some states/clients when they should not have been. This patch makes it so the current state object is kept upon any type of reload/ fetch failures. Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30 2016-05-03 15:35 +0000 [adc82a2260] Kevin Harwell * res_pjsip_outbound_publishing: After unloading the library won't load again The same thing was happening in res_pjsip_publish_asterisk. When the library was unloaded it did not unregister the object type from sorcery. Subsequent loads resulted in a failed load due to the sorcery type already existing. Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9 2016-05-03 15:39 +0000 [3b0ce5169d] Kevin Harwell * res_pjsip_outbound_publish: Won't unload if condition wait times out When res_pjsip_outbound_publish unloads it has to wait for all current publishing objects to get done. However if the wait condition times out then it does not fail the unload. This sometimes results in an infinite loop check while unloading. This patch now fails the unload operation if the condition times out. Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec 2016-05-03 14:59 +0000 [41fccbfeb1] Kevin Harwell * res_pjsip_outbound_publish: Ref leak in off nominal callback paths There were a few spots where the client object's reference was being leaked in sip_outbound_publish_callback. This patch cleans up those leaks. Change-Id: I485d0bc9335090f373026f77c548042e258461df 2016-05-03 15:31 +0000 [dfbb03cc8e] Kevin Harwell * res_pjsip_outbound_publish: Potential crash due to off nominal path It was possible for the explicit publish destroy function to be called without the pjsip client ever being initialized. This fix checks to make sure there is a client to destroy before attempting. Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c 2016-05-05 05:07 +0000 [17b6ba49ef] Joshua Colp * file: Ensure nativeformats remains valid for lifetime of use. It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 2016-04-15 09:32 +0000 [cc4c5f5693] Alexei Gradinari * res_pjsip: improve realtime performance This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05 2016-05-02 16:52 +0000 [92f85fe766] Alexei Gradinari * res_fax/t38_gateway: Peer V.21 session is created on wrong channel The channel and peer V.21 sessions are created on the same channel now. The peer V.21 session should be created only on peer channel when one of channel can handle T.38. Also this patch enable debug for T.38 gateway session if global fax debug enabled. ASTERISK-25982 Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e 2016-05-04 16:11 +0000 [4df48581f1] Alexei Gradinari * pjsip: Added "reg_server" to contacts (fixed alembic) ASTERISK-25931 Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549 2016-05-04 03:17 +0000 [02f4ca1079] Chris Trobridge * config_options.c: Expand #ifdef to contain whole if statement. ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 2016-05-02 16:08 +0000 [380ac201ac] Alexei Gradinari * res_fax: add FAXMODE variable The app_fax set FAXMODE variable, but res_fax missing this feature. This patch add FAXMODE variable which is set to either "audio" or "T38". ASTERISK-25980 Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b 2016-05-02 05:56 +0000 [0c9faaee47] Jean Aunis * app_chanspy: fix audiohook options in non read-only mode When option 'o' was not set, ChanSpy created its audiohook with the flag AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one direction only. ASTERISK-25866 #close Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0 2016-04-07 16:33 +0000 [a4cfcda036] Alexei Gradinari * res_pjsip/AMI: add contact.updated event With the old SIP module AMI sends PeerStatus event on every successfully REGISTER requests, ie, on start registration, update registration and stop registration. With PJSIP AMI sends ContactStatus only when status is changed. Regarding registration: on start registration - Created on stop registration - Removed but on update registration nothing This patch added contact.updated event. ASTERISK-25904 Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f 2016-04-30 17:52 +0000 [e61716b774] gtjoseph * pjproject_bundled: Various fixes discovered during testing of OSes For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c 2016-03-17 14:29 +0000 [080c6216b6] Andrew Nagy * app_voicemail: always copy dynamic struct to avoid race condition Voicemail email addresses can be corrupt or voicemail emails can end up being sent to the wrong email address if asterisk is reading voicemail.conf during a reload and processing an email at the same time. This patch always copies the struct that would otherwise only be copied once. ASTERISK-24463 #close Reported by: John Campbell Tested by: Etienne Lessard Tested by: Andrew Nagy Change-Id: I3a0643813116da84e2617291903d0d489b7425fb 2016-04-15 14:26 +0000 [2b1edee772] Alexei Gradinari * pjsip: Added "reg_server" to contacts. If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8 2016-05-01 02:21 +0000 [bf13b59062] Diederik de Groot * configs/basic-pbx/asterisk.conf: contains incorrect path separator Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004 2016-04-27 17:19 +0000 [2c46063d54] Richard Mudgett * res_pjsip_exten_state: Create PUBLISH messages. Create PUBLISH messages to update a third party when an extension state changes because of either a device or presence state change. A configuration example: [exten-state-publisher] type=outbound-publish server_uri=sip:instance1@172.16.10.2 event=presence ; Optional regex for context filtering, if specified only extension state ; for contexts matching the regex will cause a PUBLISH to be sent. @context=^users ; Optional regex for extension filtering, if specified only extension ; state for extensions matching the regex will cause a PUBLISH to be sent. @exten=^[0-9]* ; Required body type for the PUBLISH message. ; ; Supported values are: ; application/pidf+xml ; application/xpidf+xml ; application/cpim-pidf+xml ; application/dialog-info+xml (Planned support but not yet) @body=application/pidf+xml The '@' extended variables are used because the implementation can't extend the outbound publish type as it is provided by the outbound publish module. That means you either have to use extended variables, or implement some sort of custom extended variable thing in the outbound publish module. Another option would be to refactor that stuff to have an option which specifies the use of an alternate implementation's configuration and then have that passed to the implementation. JColp opted for the extended variables method originally. ASTERISK-25972 #close Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca 2016-04-26 16:10 +0000 [0b5292525c] Richard Mudgett * res_pjsip_exten_state: Check if body generator is available. When starting the extension state publishers, check if the requested message body generator is available. If not available give error message and skip starting that publisher. * res_pjsip_pubsub.c: Create new API if type/subtype generator registered. * res_pjsip_exten_state.c: Use new body generator API for validation. ASTERISK-25922 Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c 2016-04-28 11:35 +0000 [369182d084] Richard Mudgett * res_pjsip: Start body generator users after suppliers. Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb 2016-04-28 16:06 +0000 [3af83ea2fb] Richard Mudgett * res_pjsip_pubsub.c: Add useful information to some messages. Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a 2016-04-26 15:58 +0000 [8e1b663b87] Richard Mudgett * res_pjsip_pubsub.c: Fix body generator registration race. Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67 2016-04-28 16:54 +0000 [30415944a8] gtjoseph * pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, the input uri string will become corrupted if it contains escape sequences. It's not possible to automatically strdup or strdupa the input string because the output uri pj_str_t's will have pointers to chunks of the input string. Getting around this would require more memory management code and wouldn't be worth the savings of doing the unescape in place. ASTERISK-25970 #close Reported-by: Dmitriy Serov Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88 2016-04-26 15:13 +0000 [906ea2c43f] Richard Mudgett * res_pjsip_pubsub.h: Fix doxygen association. Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632 2016-04-25 16:00 +0000 [76ea4cfaae] Richard Mudgett * res_pjsip_outbound_publish.c: Remove redundant flag check. Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353 2016-03-07 18:34 +0000 [4ebf9a938d] gtjoseph * res_pjsip: Add ability to identify by Authorization username A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d 2016-04-27 13:23 +0000 [2b150f0b80] Mark Michelson * func_odbc: Check connection status before executing queries. A recent change to func_odbc made it so that a single connection was maintained per DSN. The problem was that the code was optimistic about the health of the connection after initially opening it and did nothing to re-connect in case the connection had died. This change adds a check before executing a query to ensure that the connection to the database is still up and running. ASTERISK-25963 #close Reported by Ross Beer Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d 2016-04-15 11:59 +0000 [860b135c88] Alexei Gradinari * res_pjsip: disable multi domain to improve realtime performace This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7 2016-04-01 07:50 +0000 [7281770710] Jean Aunis * app_chanspy: reduce audio loss on the spying channel. ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when queues grow too large or when read and write queues go out of sync. Now these flags are set conditionally: - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not be set on the audiohook ASTERISK-25866 Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd 2016-04-14 07:03 +0000 [81ea80b74c] Joshua Colp * res_pjsip_exten_state: Add config support for exten state publishers. This change adds the ability to configure outbound publishing of extension state. Right now stuff is merely set up to store the configuration and to register a global extension state callback. The act of constructing the body and sending is not yet complete. Configurable elements right now are a regex for filtering the context, a regex for filtering the extension, and the body type to publish. ASTERISK-25922 #close Change-Id: Ia7e630136dfc355073c1cadff8ad394a08523d78 2016-04-26 11:13 +0000 [c480159045] Joshua Colp * chan_sip: Give more time for TCP/TLS threads to stop. The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6 2016-04-26 05:48 +0000 [8ae69cffef] Joshua Colp * app_queue: Fix crash when unloading module. When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b 2016-04-24 22:51 +0000 [284bb814ac] gtjoseph * config: Fix ast_config_text_file_save2 writability check for missing files A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 2016-04-25 08:11 +0000 [f99ec857c8] Javier Acosta * Fix case sensitive actions in AMI QueueSummary and QueueStatus ASTERISK-25954 #close Reported by: Javier Acosta Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 (cherry picked from commit c0688a6398f27296ff849848a2e416e036d794e3) 2016-04-21 14:23 +0000 [30ab21d5fa] Kevin Harwell * app_queue: queue members can receive multiple calls It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 2016-04-22 17:53 +0000 [99fcf2a791] gtjoseph * res_agi: Prevent run_agi from eating frames it shouldn't The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645 2016-04-22 15:25 +0000 [757ec6172b] Richard Mudgett * test_message.c: Wait longer in case dialplan also processes the test message. Bumped the wait from 1 second to 5 seconds. The test message was hitting my default call handler and failing the test because it took longer. Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba 2016-04-21 23:53 +0000 [41ecf22587] Kirill Katsnelson * chan_sip: Make autocreated peers send PeerStatus events Since Stasis has been introduced, an attempt to send AMI messages by an autocreated peer caused a crash, and all events from autocreated peers were semi-inadvertently disabled altogether in 0b83761. This change restores the disabled functionality. ASTERISK-25950 Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974 2016-04-13 17:09 +0000 [b3cc74fda9] Richard Mudgett * manager_channels.c: Fix allocation failure crash. An earlier allocation failure failed to create a channel snapshot for the AMI HangupRequest/SoftHangupRequest event which resulted in a crash in channel_hangup_request_cb(). Where the stasis message gets generated cannot tell if the NULL snapshot returned was because of an allocation failure or the channel was a dummy channel. * Made channel_hangup_request_cb() check if the channel blob has a snapshot and exit if it doesn't. * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 2016-04-13 13:50 +0000 [a63656b419] Richard Mudgett * Bridge system: Fix memory leaks and double frees on impart failure. You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 2016-04-13 13:20 +0000 [71dfa35540] Richard Mudgett * bridge_softmix.c: Fix crash if channel fails to join mixing tech. softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b 2016-04-12 15:29 +0000 [06632a0d11] Richard Mudgett * Manager: Short circuit AMI message processing. Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 2016-04-13 17:54 +0000 [6ddd856b86] Richard Mudgett * manager.c: Eliminate most RAII_VAR usage. * Made ast_manager_event_blob_create() not allocate the ao2 event object with a lock as it is not needed. Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c 2016-04-22 13:49 +0000 [924738e950] Mark Michelson * func_odbc: Use one connection per DSN. res_odbc was changed in Asterisk 13.8.0 to remove connection management, opting instead to let unixodbc maintain open connections and return those to Asterisk as requested. This was a boon for realtime, since it meant that multiple threads could potentially run parallel queries since they could each be using their own database connections. However, on the user-facing side, func_odbc, there were some inherent behaviors being relied on that no longer hold true after the change. One such reported behavior was that MySQL's LAST_INSERTED_ID() works per-connection. This means that if Asterisk uses separate connections for every database operation, whereas before it used one connection for everything, we have broken expectations and functionality. The fix provided in this patch is to make func_odbc use a single database connection per DSN. This way, user-facing database usage will have the same behavior as it did pre-13.8.0. However, realtime, which is the real workhorse of database interaction, will continue to let unixodbc manage connections. ASTERISK-25938 #close Reported by Edwin Vandamme Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc 2016-04-22 13:02 +0000 [6ede210c98] Leif Madsen * Remove reference to non-existent sip.conf option Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8 2016-04-21 08:26 +0000 [c991e5472e] Diederik de Groot * lock.c: Check *lt before dereferencing it *lt is NULL if t->tracking == 0 ASTERISK-25948 #close Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba 2016-04-15 14:36 +0000 [6b1a632290] Richard Mudgett * res_stasis: Handle re-enter stasis bridge with swap channel. We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee 2016-04-19 16:58 +0000 [1c5248c383] Richard Mudgett * bridge: Hold off more than one imparting channel at a time. An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 2016-04-19 17:52 +0000 [70e860ec49] gtjoseph * res_pjsip_callerid: Clear out display name if id->name is not valid When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning the From header, then it overwrites the display name and uri from the channel's connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was leaving the display name from the From header in the new RPID or PAI header. On an attended transfer where the originator had a caller id number set but not a display name, the re-INVITE to the final transferee had the number of the originator but the display name of the transferer. Added a check to clear out the display name in the new header if connected.id.name was invalid. ASTERISK-25942 #close Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b 2016-04-19 13:02 +0000 [d95512a7dd] Joshua Colp * app_talkdetect: Make the module core supported. This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 2016-04-18 12:12 +0000 [0235a66532] Mark Michelson * PJSIP: Remove PJSIP parsing functions from uri length validation. The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 2016-04-18 17:00 +0000 [b8b60135ec] Mark Michelson * res_pjsip_registrar: Fix bad memory-ness with user_agent. Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 2016-04-18 13:41 +0000 [6cfa02394f] Joshua Colp * res_pjsip_transport_management: Allow unload to occur. At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a 2016-04-15 11:41 +0000 [6365f0018f] Richard Mudgett * bridge_channel.c: Ignore role setup failure in channel push. We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 2016-04-17 15:37 +0000 [f06ce7f90a] Jaco Kroon * chan_sip: Don't verify table if rtupdate=no If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d 2016-04-18 04:53 +0000 [dbb47e0a47] ibercom * app_queue: Frequent segfaults in function can_ring_entry() ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117 2016-04-15 16:51 +0000 [af114edb8b] Richard Mudgett * stasis_bridge.c: Update stasis bridge push diagnostic messages. Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a 2016-04-12 14:55 +0000 [5e64d7e7a3] Mark Michelson * Dial: Combine frame handling functions. There is a good amount of repetition in the two frame handling routines in the Dial API. This commit combines the two functions into one. This is in preparation for an upcoming commit that adds the ability to handle frames for a channel in a bridge. ASTERISK-25925 Reported by Mark Michelson Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58 2016-04-11 16:20 +0000 [a6e2ba187a] Alexei Gradinari * Codecs: strip codec name while parsing allow/disallow options Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d 2016-04-14 13:49 +0000 [be4333ddad] Mark Michelson * transport management: Register thread with PJProject. The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 2016-03-17 12:28 +0000 [e83499df56] gtjoseph * res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c 2016-03-08 12:12 +0000 [216f22fd0f] Mark Michelson * res_pjsip_transport_management: Kill idle TCP connections. "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 2016-03-08 10:52 +0000 [d9fba46016] Mark Michelson * Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433 2016-04-14 07:23 +0000 [7b8b6e2e4f] Mark Michelson * AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d 2016-04-12 13:10 +0000 [ff3af764de] Richard Mudgett * bridge_softmix.c: Fix crash if could not allocate the dsp. Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372 2016-04-13 13:38 +0000 [caa416d5f3] gtjoseph * stringfields: Update extended string fields for master only. In 13, the new ast_string_field_header structure had to be dynamically allocated and assigned to a pointer in ast_string_field_mgr to preserve ABI compatability. In master, it can be converted to being a structure-in-place in ast_string_field_mgr to eliminate the extra alloc and free calls. Change-Id: Ia97c5345eec68717a15dc16fe2e6746ff2a926f4 2016-04-12 15:41 +0000 [bd3671b397] gtjoseph * pjproject: Add patch for removing strip of '[]' from header params From the patch submitted to Teluu on 4/12/2016 <<<<<<<<< The wholesale stripping of '[]' from header parameters causes issues if something (like a port) occurs after the final ']'. '[2001:a::b]' will correctly parse to '2001:a::b' '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left with ':8080' and parsing stops with a syntax error. I can't even find a case where stripping the '[]' is a good thing anyway. Even if you continued to parse and resulted in a string that looks like this... '2001:a::b:8080', it's not valid. This came up in Asterisk because Kamailio sends us a Contact with an alias URI parameter that has an IPv6 address in it like this: Contact: which should be legal but causes a syntax error because of the characters after the final ']'. Even if it didn't, the '[]' should still not be stripped. I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 enabled. No issues were caused by removing the code that strips the '[]'. >>>>>>>>>>> ASTERISK-25123 #close Reported-by: Anthony Messina Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a 2016-04-12 09:10 +0000 [5a0534dc62] Joshua Colp * app_voicemail: Fix test_voicemail_notify_endl test. The test_voicemail_notify_endl test checks the end-of-line characters of an email message to confirm that they are consistent. The test wrongfully assumed that reading from the email message into a buffer will always result in more than 1 character being read. This is incorrect. If only 1 character was read the test would go outside of the buffer and access other memory causing a crash. The test now checks to ensure that 2 or more characters are read in ensuring the test stays within the buffer. ASTERISK-25874 #close Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710 2016-04-07 12:02 +0000 [c00c298a0e] Alexei Gradinari * app_voicemail/IMAP: function 'save_to_folder' creates wrong folder If try to move message to Cust1 (number 5) the function 'save_to_folder' tries to create Greeting folder instead of Cust1. This patch fixed it by setting GREETINGS_FOLDER = -1 ASTERISK-24927 #close Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 2016-04-07 16:18 +0000 [49813bc9e5] Alexei Gradinari * res_pjsip: Add headers to AMI Event ContactStatusDetail * Added Useragent and RegExpire headers to AMI Event ContactStatusDetail with associated documentation. ASTERISK-25903 #close Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239 2016-04-05 16:56 +0000 [4e00e31ef1] Alexei Gradinari * res_pjsip_outbound_publish: Add transport for outbound PUBLISH The first available transport of the appropriate type is used now. This patch adds new config option 'transport' for outbound-publish. If transport is set then outbound PUBLISH requests will use this transport. ASTERISK-25901 #close Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151 2016-04-11 14:26 +0000 [2cc56573de] Jaco Kroon * core_unreal: Fix hangupcauses not getting set on Local channels ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa 2016-04-01 13:30 +0000 [a621dd5e96] gtjoseph * res_pjsip contact: Lock expiration/addition of contacts Contact expiration can occur in several places: res_pjsip_registrar, res_pjsip_registrar_expire, and automatically when anyone calls ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar may also be attempting to renew or add a contact. Since none of this was locked it was possible for one thread to be renewing a contact and another thread to expire it immediately because it was working off of stale data. This was the casue of intermittent registration/inbound/nominal/multiple_contacts test failures. Now, the new named lock functionality is used to lock the aor during contact expire and add operations and res_pjsip_registrar_expire now checks the expiration with the lock held before deleting the contact. ASTERISK-25885 #close Reported-by: Josh Colp Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059 2016-04-10 14:16 +0000 [8637f29d24] gtjoseph * pjproject: Add patch to fix Via IPv6 parsing There's a bug in pjproject's sip_parser where the ":" wasn't correctly interpreted. This is causing IPv6 addresses in the "received" parameter of the Via header to cause a syntax check failure. This patch was submitted to Teluu on 4/10/2016. ASTERISK-25910 #close Reported-by: Anthony Messina Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922 2016-03-31 20:04 +0000 [216abb0ae7] gtjoseph * lock: Add named lock capability Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 2016-04-07 11:37 +0000 [f9dab80816] Alexei Gradinari * app_voicemail/IMAP: IMAP access FATAL error: Out of memory Sometimes uw-imap function 'mail_fetchbody' returns huge len which then pass to uw-imap function 'rfc822_base64'. uw-imap tries to allocate huge memory and abort() on fail. This patch check the len. If the len more than max size (128 Mbytes) log error. This patch also set variables len, newlen to avoid uninizialezed len. This patch also check pointer returned by rfc822_base64. ASTERISK-25899 #close Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca 2016-04-07 16:39 +0000 [b3be945415] Alexei Gradinari * res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event BLF pickup isn't working on Cisco SPA and Snom phones if the direction="recipient" attribute is missing in 'dialog' tag. This patch adds direction="recipient" if extension state is Ringing. ASTERISK-24601 #close Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c 2016-04-06 17:57 +0000 [6138a75e8e] Richard Mudgett * pbx.h: Make ast_state_cb_type take more const. This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572 2016-04-07 10:59 +0000 [72c19f7dc5] Richard Mudgett * pbx.c: Minor code rearangements. * Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 2016-04-07 12:26 +0000 [28cefc3e88] Richard Mudgett * pbx: Update doxygen for extension state watchers. Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e 2016-04-07 11:49 +0000 [751d7a5a49] gtjoseph * alembic: Remove batch operations (and sqlite support) Because SQLite doesn't support full ALTER capabilities, alembic scripts require batch operations. However, that capability wasn't available until 0.7.0 which some distributions haven't reached yet. Therefore, the batch operations introduced in commit 86d6e44cc (review 2319) have been reverted and SQLite is unsupported again, for now anyway. Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. ASTERISK-25890 #close Reported-by: Harley Peters Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80 2016-04-07 11:05 +0000 [2eaeea690d] Joshua Colp * res_pjsip_registrar_expire: Fix race condition at shutdown. When shutting down, the PJSIP sorcery is destroyed. The registrar expiration module queries the PJSIP sorcery to determine what to expire. As there was no synchronization between termination of the expiration thread and the unloading of the module it was possible for the thread to try to access the PJSIP sorcery after it had been destroyed. This change ensures that the thread is shut down before allowing the module to be considered unloaded. Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b 2016-04-06 16:28 +0000 [3e5672d843] Joshua Colp * res_pjsip: Fix configuration setting of "regcontext". Due to a merge problem two options were swapped causing the regcontext setting to not get set. Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1 2016-04-06 08:01 +0000 [8ed5f61152] Jacek Konieczny * frame.c: Copy the whole subclass in ast_frdup(). The problem is ast_frdup() does not copy whole frame.subclass for voice, video and image frames, only the format is copied. For video frames, the subclass structure contains the .frame_ending flag used to put the RTP marker where it needs to be. ASTERISK-25894 #close Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33 2016-03-30 17:18 +0000 [abbb2edd4c] Mark Michelson * ARI: Add method to Dial a created channel. This adds a new ARI method that allows for you to dial a channel that you previously created in ARI. By combining this with the create method for channels, it allows for a workflow where a channel can be created, manipulated, and then dialed. The channel is under control of the ARI application during all stages of the Dial and can even be manipulated based on channel state changes observed within an ARI application. The overarching goal for this is to eventually be able to add a dialed channel to a Stasis bridge earlier than the "Up" state. However, at the moment more work is needed in the Dial and Bridge APIs in order to facilitate that. ASTERISK-25889 #close Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205 2016-03-30 17:01 +0000 [dd48d60c5b] Mark Michelson * ARI: Add method to create a new channel. This adds a new ARI method to the channels resource that allows for the creation of a new channel. The channel is created and then placed into the specified Stasis application. This is different from the existing originate method that creates a channel, dials it, and then places the answered channel into the dialplan or a Stasis application. This method does not attempt to call the channel at all. Dialing is left as a later step after channel creation. This allows for pre-dialing channel manipulation if desired. ASTERISK-25889 Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8 2016-03-28 11:31 +0000 [1dc5e28624] Joshua Colp * pbx: Add support for autohints. This change introduces the concept of autohints. These are hints which are created as a result of device state changes occurring within the core. When this happens a hint will be created (if it does not exist already) using the device name as the extension. For example if a device state change is received for "PJSIP/bob" and autohints are enabled on a context then a hint will exist in that context for "bob" with a device of "PJSIP/bob". For virtual or custom device states the name after the type will be used. For example if the device state of "Custom:bob" changes then a hint will exist in that context for "bob" with a device of "Custom:bob". This functionality can be enabled in extensions.conf by placing "autohints=yes" in a context. ASTERISK-25881 #close Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e 2016-04-05 14:23 +0000 [a098251e7e] Mark Michelson * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. At this point, PJMedia's SDP negotiator saves Asterisk's local state as being recvonly. Now, when the device attempts to unhold, it again uses a deferred SDP reinvite, so we end up doing the following: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating recvonly on streams * Device sends ACK with SDP indicating sendonly on streams The problem here is that Asterisk offered recvonly, and by RFC 3264's rules, if an offer is recvonly, the answer has to be sendonly. The result is that the device is not taken off hold. What is supposed to happen is that Asterisk should indicate sendrecv in the 200 OK that it sends. This way, the device has the freedom to indicate sendrecv if it wants the stream taken off hold, or it can continue to respond with sendonly if the purpose of the reinvite was something else (like a session timer refresher). The fix here is to alter the SDP negotiator's state when we receive a reinvite with no SDP. If the negotiator's state is currently in the recvonly or inactive state, then we alter our local state to be sendrecv. This way, we allow the device to indicate the stream state as desired. ASTERISK-25854 #close Reported by Robert McGilvray Change-Id: I7615737276165eef3a593038413d936247dcc6ed 2016-03-30 16:47 +0000 [ef4d3f1328] Mark Michelson * Dial: Add function to append already-created channel. The Dial API takes responsiblity for creating an outbound channel when calling ast_dial_append(). This commit adds a new function, ast_dial_append_channel(), which allows us to create the channel outside the Dial API and then to append the channel to the ast_dial structure. This is useful for situations where the channel's creation and dialing are distinct operations. Upcoming ARI early bridge work will illustrate its usage. ASTERISK-25889 Change-Id: Id8179f64f8f99132f80dead8d5db2030fd2c0509 2016-03-27 23:33 +0000 [984d6fd95c] gtjoseph * config: Allow filters when appending to a category In sorcery based config files where there are multiple categories with the same name, you can't use the (+) operator to reliably append to a category because config.c stops looking when it finds the first one with the same name. Example: [1000] type = endpoint [1000] type = aor [1000](+) authenticate_qualify = yes This config will fail because config.c appends authenticate_qualify to the first category it finds, the endpoint, and that's not valid for endpoint. Solution: The capability to find a category that contains a certain variable already exists so the only real change was to parse anything after the '+' that's not a comma, as a filter string. [1000] type = endpoint [1000] type = aor [1000](+type=aor) authenticate_qualify = yes This now works as expected. Although the following example doesn't make any sense for pjsip, you can even specify multiple filters: [1000](+type=aor&qualify_frequency=10) ASTERISK-25868 #close Reported-by: Nick Repin Change-Id: I10773da4c79db36fbf1993961992af63d3441580 2016-04-05 10:21 +0000 [784fb43f43] Joshua Colp * res_http_websocket: Make core supported. Websockets are a core part of ARI support and as such this module should also be core supported. Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c 2016-03-25 23:22 +0000 [4d40b161c3] gtjoseph * stringfields: Refactor to allow fields to be added to the end of structures String fields are great, except that you can't add new ones without breaking ABI compatibility because it shifts down everything else in the structure. The only alternative is to add your own char * field to the end of the structure and manage the memory yourself which isn't ideal, especially since you then can't use the OPT_STRINGFIELD_T type. Background: The reason string fields had to be declared inside the AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS declared the pool, then the fields, then the manager, you could use the offsets of the pool and manager and iterate over the sequential addresses in between to access the fields. The actual pool, field allocation and field set operations don't actually care where the field is. It's just iteration over the fields that was the problem. Solution: Extended String Fields An extended string field is one that is declared outside the AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent structure. Other than using AST_STRING_FIELD_EXTENDED instead of AST_STRING_FIELD, it looks the same as other string fields. It's storage comes from the pool and it participates in string field compare and copy operations peformed on the parent structure. It's also a valid target for the OPT_STRINGFIELD_T aco option type. Implementation: To keep track of the extended fields and make sure that ABI isn't broken, the existing embedded_pool pointer in the manager structure was repurposed to be a pointer to a separate header structure that contains the embedded_pool pointer plus a vector of fields. The length of the manager structure didn't change and the embedded_pool pointer isn't used in the macros, only the stringfields C code. A side benefit of this is that changing the header structure in the future won't break ABI. ast_string_fields_init initializes the normal string fields and appends them to the vector, and subsequent calls to ast_string_field_init_extended initialize and append the extended fields. Cleanup, ast_string_fields_cmp, and ast_string_fields_copy can now work on the vector instead of sequentially traversing the addresses between the pool and manager. The total size of a structure using string fields didn't change, whether using extended fields or not, nor have the offsets of any structure members, either inside the original block or outside. Adding an extended field to the end of a structure is the same as adding a char *. Details: The stringfield C code was pulled out from utils.c and into stringfields.c. It just made sense. Additional work was done in ast_string_field_init and ast_calloc_with_stringfields to handle the allocation of the new header structure and the vector, and the associated cleanup. In the process some additional NULL pointer checking was added. A lot of work was done in stringfields.h since the logic for compare and copy is there. Documentation was added as well as somne additional NULL checking. The ability to call ast_calloc_with_stringfields with a number of structures greater than 1 never really worked. Well, the calloc worked but there was no way to access the additional structures or clean them up. It was agreed that there was no use case for requesting more than 1 structure so an ast_assert was added to prevent it and the iteration code removed. Testing: The stringfield unit tests were updated to test both normal and extended fields. Tests for ast_string_field_ptr_set_by_fields and ast_calloc_with_stringfields were also added. As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except res_pjsip itself, saved off. The patch was then added and a full compile and install was performed. Then the older res_pjsip_* moduled were copied over the installed versions so res_pjsip was new and the rest were old. No issues. contact->aor, which is a char * at the end of contact, was then changed to an extended string field and a recompile and reinstall was performed, again leaving stock versions of the the res_pjsip_* modules. Again, no issues with the res_pjsip_* modules using the old stringfield implementation and with contact->aor as a char *, and res_pjsip itself using the new stringfield implementation and contact->aor being an extended string field. Finally, several existing string fields were converted to extended string fields to test OPT_STRINGFIELD_T. Again, no issues. Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61 2016-04-04 18:02 +0000 [c07e1190ec] gtjoseph * res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7 I forgot the new voicemail_extension wasn't a stringfield and didn't check for NULL where I should have. Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb 2016-04-03 11:47 +0000 [060b7b83bc] gtjoseph * install_prereq: Fix check_installed_debs remove subversion check_installed_debs wasn't handling virtual packages like libsrtp-dev and libresample-dev and on multiarch systems it was accidentally filtering out all packages if any :i386 packages were found instead of just filtering out the :i386 packages themselves. Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda 2016-04-01 13:09 +0000 [433d2c4bbf] gtjoseph * utils.c: Fix typo in handle_show_locks ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e). Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf 2016-03-30 18:34 +0000 [304f81780d] gtjoseph * pjproject_bundled: Fix use of LDCONFIG for shared library link creation LDCONFIG apparently isn't set to something sane on all systems so the creation of the shared library links fails. Instead of just testing for non-blank, main/Makefile now checks that LDCONFIG is actually executable and reverts to LN if it isn't. This applies to both libasteriskpj and libasteriskssl. Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG. ASTERISK-25873 #close Reported-by: Hans van Eijsden Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729 2016-03-29 13:47 +0000 [0ea742d33a] Richard Mudgett * res_stasis: Add control ref to playback and recording structs. The stasis_app_playback and stasis_app_recording structs need to have a struct stasis_app_control ref. Other threads can get a reference to the playback and recording structs from their respective global container. These other threads can then use the control pointer they contain after the control struct has gone. * Add control ref to stasis_app_playback and stasis_app_recording structs. With the refs added, the control command queue can now have a circular control reference which will cause the control struct to never get released if the control's command queue is not flushed when the channel leaves the Stasis application. Also the command queue needs better protection from adding commands if the control->is_done flag is set. * Flush the control command queue on exit. ASTERISK-25882 #close Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d 2016-03-28 18:10 +0000 [53f63ad770] Richard Mudgett * res_stasis: Fix crash on a hanging up channel. * Give the struct stasis_app_control ao2 object a ref to the channel held in the object. Now the channel will still be around if a thread needs to post a stasis message instead of crash because the topic was destroyed. * Moved stopping any lingering silence generator out of the struct stasis_app_control destructor and made it a part of exiting the Stasis application. Who knows which thread the destructor will be called under so it cannot affect the channel's silence generator. Not only was the channel unprotected when the silence generator was stopped, stasis may no longer even control the channel. ASTERISK-25882 Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4 2016-03-30 13:31 +0000 [2fab4d7da8] Richard Mudgett * res_stasis.c: Protect channel datastore list from stasis end. Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95 2016-03-29 18:06 +0000 [ece2edaa04] Richard Mudgett * res_ari: Cannot get control also means channel is unavailable. The only caller of ari_bridges_play_found() has this note: If ari_bridges_play_found fails because the channel is unavailable for playback, The channel will be removed from the playback list soon. We can keep trying to get channels from the list until we either get one that will work or else there isn't a channel for this bridge anymore, in which case we'll revert to ari_bridges_play_new. Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6 2016-03-29 14:29 +0000 [2f36cba4b5] Richard Mudgett * res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name(). Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7 2016-03-28 14:23 +0000 [34457dd9db] Richard Mudgett * core_unreal.c: Add clarification comment about channel ref. Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3 2016-03-30 12:38 +0000 [2b3261cd36] gtjoseph * res_pjsip_mwi: Allow subscribe to vm access extension as an alias Background: If your extension is 1000 and the voicemail access extension is 1571 and you dial 1571, usually a dialplan rule calls voicemailmain with your extension and you are placed directly in your mailbox. Therefore most admins program the voicemail (or other speed dial) button on their phones to the access extension. Some phones (Snom at least) use whatever is programmed there to also subscribe for MWI and so can't dial one number and subscribe to another. This works fine in chan_sip because chan_sip completely ignores the user portion of the SUBSCRIBE message request URI. If it can match the peer, is subscribes to the peer's mailbox. The user could be set to anything or nothing and you'd still get subscribed to your mailbox. Issue: chan_pjsip actually uses the user portion of the URI to find an aor and its mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can create an aor for 1571 but you certainly can't add your entire voicemail system's mailboxes to it and everyone would get notified of every MWI. Solution: When an MWI subscribe comes in and an aor can't be found that matches the resource directly, check the resource against the endpoint's aors. If an aor is found that has a voicemail_extension that matches the resource, use it. ASTERISK-25865 Reported-by: Ross Beer Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e 2016-03-24 22:55 +0000 [e2524fcee3] gtjoseph * res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea 2016-03-30 09:46 +0000 [724b9ab28f] gtjoseph * res_rtp_asterisk: Fix placement of txcount increment Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount for rtcp packets as well as rtp packets and that was causing sender reports to be generated instead of receiver reports in cases where no rtp was actually being sent. Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, to rtp_sento which only handles rtp packets. Discovered by the hep/rtcp-receiver test. Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5 2016-03-26 22:33 +0000 [c4064727d2] gtjoseph * chan_pjsip: Add 'pjsip show channelstats' Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869 2016-03-25 10:59 +0000 [970803efcb] Jacek Konieczny * res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 explicitly states: There MUST be a separate DTLS-SRTP session for each distinct pair of source and destination ports used by a media session This means RTP keying material cannot be used for DTLS RTCP, which was the reason why RTCP encryption would fail. ASTERISK-25642 Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a 2016-03-25 10:42 +0000 [9785e8d090] Jacek Konieczny * app_echo: forward and generate VIDUPDATE frames When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e 2016-03-27 12:53 +0000 [44ffb5105a] gtjoseph * res_rtp_asterisk: Fix packet stats on bridged connection rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated for bridged streams because the calulations were being done after the bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. Moved the calculations so they occur for all valid received packets and all transmitted packets. Also added rxoctetcount and txoctetcount to ast_rtp_instance_stat. Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb 2016-03-10 19:52 +0000 [c971a64366] gtjoseph * res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS No one seemed to notice but every time an OPTIONS goes out, it goes out with a From of "asterisk" (or whatever the default from_user is set to), even if you specify an endpoint. The issue had several causes... qualify_contact is only called with an endpoint if called from the CLI. If the endpoint is NULL, qualify_contact only looks up the endpoint if authenticate_qualify=yes. Even then, it never passes it on to ast_sip_create_request where the From header is set. Therefore From is always "asterisk" (or whatever the default from_user is set to). Even if ast_sip_create_request were to get an endpoint, it only sets the From if endpoint->from_user is set. The fix is 4 parts... First, create_out_of_dialog_request was modified to use the endpoint id if endpoint was specified and from_user is not set. Second, qualify_contact was modified to always look up an endpoint if one wasn't specified regardless of authenticate_qualify. It then passes the endpoint on to create_out_of_dialog_request. Third (and most importantly), find_an_endpoint was modified to find an endpoint by using an "aors LIKE %contact->aor%" predicate with ast_sorcery_retrieve_by_fields. As such, this patch will only work if the sorcery realtime optimizations patch goes in. Otherwise we'd be pulling the entire endpoints database every time we send an OPTIONS. Since we already know the contact's aor, the on_endpoint callback was also modified to just check if the contact->aor is an exact match to one of the endpoint's. Finally, since we now have an endpoint for every OPTIONS request, res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was updated to get the transport from the endpoint and set it on tdata. Now the correct transport is used. Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af 2016-03-08 15:55 +0000 [c948ce9651] gtjoseph * sorcery/res_pjsip: Refactor for realtime performance There were a number of places in the res_pjsip stack that were getting all endpoints or all aors, and then filtering them locally. A good example is pjsip_options which, on startup, retrieves all endpoints, then the aors for those endpoints, then tests the aors to see if the qualify_frequency is > 0. One issue was that it never did anything with the endpoints other than retrieve the aors so we probably could have skipped a step and just retrieved all aors. But nevermind. This worked reasonably well with local config files but with a realtime backend and thousands of objects, this was a nightmare. The issue really boiled down to the fact that while realtime supports predicates that are passed to the database engine, the non-realtime sorcery backends didn't. They do now. The realtime engines have a scheme for doing simple comparisons. They take in an ast_variable (or list) for matching, and the name of each variable can contain an operator. For instance, a name of "qualify_frequency >" and a value of "0" would create a SQL predicate that looks like "where qualify_frequency > '0'". If there's no operator after the name, the engines add an '=' so a simple name of "qualify_frequency" and a value of "10" would return exact matches. The non-realtime backends decide whether to include an object in a result set by calling ast_sorcery_changeset_create on every object in the internal container. However, ast_sorcery_changeset_create only does exact string matches though so a name of "qualify_frequency >" and a value of "0" returns nothing because the literal "qualify_frequency >" doesn't match any name in the objset set. So, the real task was to create a generic string matcher that can take a left value, operator and a right value and perform the match. To that end, strings.c has a new ast_strings_match(left, operator, right) function. Left and right are the strings to operate on and the operator can be a string containing any of the following: = (or NULL or ""), !=, >, >=, <, <=, like or regex. If the operator is like or regex, the right string should be a %-pattern or a regex expression. If both left and right can be converted to float, then a numeric comparison is performed, otherwise a string comparison is performed. To use this new function on ast_variables, 2 new functions were added to config.c. One that compares 2 ast_variables, and one that compares 2 ast_variable lists. The former is useful when you want to compare 2 ast_variables that happen to be in a list but don't want to traverse the list. The latter will traverse the right list and return true if all the variables in it match the left list. Now, the backends' fields_cmp functions call ast_variable_lists_match instead of ast_sorcery_changeset_create and they can now process the same syntax as the realtime engines. The realtime backend just passes the variable list unaltered to the engine. The only gotcha is that there's no common realtime engine support for regex so that's been noted in the api docs for ast_sorcery_retrieve_by_fields. Only one more change to sorcery was done... A new config flag "allow_unqualified_fetch" was added to reg_sorcery_realtime. "no": ignore fetches if no predicate fields were supplied. "error": same as no but emit an error. (good for testing) "yes": allow (the default); "warn": allow but emit a warning. (good for testing) Now on to res_pjsip... pjsip_options was modified to retrieve aors with qualify_frequency > 0 rather than all endpoints then all aors. Not only was this a big improvement in realtime retrieval but even for config files there's an improvement because we're not going through endpoints anymore. res_pjsip_mwi was modified to retieve only endpoints with something in the mailboxes field instead of all endpoints then testing mailboxes. res_pjsip_registrar_expire was completely refactored. It was retrieving all contacts then setting up scheduler entries to check for expiration. Now, it's a single thread (like keepalive) that periodically retrieves only contacts whose expiration time is < now and deletes them. A new contact_expiration_check_interval was added to global with a default of 30 seconds. Ross Beer reports that with this patch, his Asterisk startup time dropped from around an hour to under 30 seconds. There are still objects that can't be filtered at the database like identifies, transports, and registrations. These are not going to be anywhere near as numerous as endpoints, aors, auths, contacts however. Back to allow_unqualified_fetch. If this is set to yes and you have a very large number of objects in the database, the pjsip CLI commands will attempt to retrive ALL of them if not qualified with a LIKE. Worse, if you type "pjsip show endpoint " guess what's going to happen? :) Having a cache helps but all the objects will have to be retrieved at least once to fill the cache. Setting allow_unqualified_fetch=no prevents the mass retrieve and should be used on endpoints, auths, aors, and contacts. It should NOT be used for identifies, registrations and transports since these MUST be retrieved in bulk. Example sorcery.conf: [res_pjsip] endpoint=config,pjsip.conf,criteria=type=endpoint endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error ASTERISK-25826 #close Reported-by: Ross Beer Tested-by: Ross Beer Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67 2016-03-25 23:19 +0000 [8e8cf80cea] Philip Correia * res_parking: Fix blind transfer dynamic lots creation. Blind transfers to a recognized parking extension need to use the parker's channel variable values to create the dynamic parking lot. This is because there is always only one parker while the parkee may actually be a multi-party bridge. A multi-party bridge can never supply the needed channel variables to create the dynamic parking lot. In the multi-party bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and channel variables are inherited by the local channel used to park the bridge. * In park_common_setup(), make use the parker instead of the parkee to supply the dynamic parking lot channel variable values. In all but one case, the parkee is the same as the parker. However, in the recognized parking extension blind transfer scenario for a two party bridge they are different channels. For consistency, we need to use the parker channel. * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the local channel when blind transferring a multi-party bridge to a recognized parking extension. * When a local channel starts a call, the Local;2 side needs to inherit the CHANNEL(parkinglot) value from Local;1. The DTMF one-touch parking case wasn't even trying to create dynamic parking lots before it aborted the attempt. * In parking_park_call(), add missing code to create a dynamic parking lot. A DTMF bridge hook is documented as returning -1 to remove the hook. Though the hook caller is really coded to accept non-zero. See the ast_bridge_hook_callback typedef. * In feature_park_call(), don't remove the DTMF one-touch parking hook because of an error. ASTERISK-24605 #close Reported by: Philip Correia Patches: call_park.patch (license #6672) patch uploaded by Philip Correia Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9 2016-03-23 14:24 +0000 [3cf714031c] Richard Mudgett * res_parking: Cleanup find_channel_parking_lot_name() usage. Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02 2016-03-18 14:01 +0000 [13e75ee04f] Richard Mudgett * res_parking: Misc fixes. res/parking/parking_applications.c: * Add malloc fail checks in setup_park_common_datastore(). * Fix playing parking failed announcement to only happen on non-blind transfers in park_app_exec(). It could never go out before because a test was provedly always false. res/parking/parking_bridge.c: * Fix NULL tolerance in generate_parked_user() because bridge_parking_push() can theoretically pass a NULL parker channel if the parker channel went away for some reason. * Clarify some weird code dealing with blind_transfer in bridge_parking_push(). res/parking/parking_bridge_features.c: * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel which will be bulk copied to the Local;2 channel on the subsequent ast_call(). The additional advantage is if the parker channel has the BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed to be overridden. res/parking/parking_manager.c: * Fix AMI Park action input range checking of the Timeout header in manager_park(). * Reduced locking scope to where needed in manager_park(). res/res_parking.c: * Fix some off nominal missing unlocks by eliminating the returns. Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca 2014-12-15 05:23 +0000 [e2853ae337] Philip Correia * res_parking: Update parking documentation for dynamic parking lots. * Remove duplicate res_parking.conf courtesytone config option documentation. ASTERISK-24596 #close Reported by: Philip Correia ASTERISK-24605 Reported by: Philip Correia Patches: call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5 2016-03-25 06:02 +0000 [72a897c534] Joshua Colp * media_cache: Demote warning to debug as it may occur often. The file playback system will now query the media cache and then the old file functionality. Under normal conditions this will result in the cache failing to retrieve a file causing a warning message to get output each time a file is played back. This change demotes this warning to a debug message. Change-Id: Ib72246ba300b5cce32774bfb3c26634bfb708624 2016-03-10 16:58 +0000 [89e94e886c] Mark Michelson * Restrict CLI/AMI commands on shutdown. During stress testing, we have frequently seen crashes occur because a CLI or AMI command attempts to access information that is in the process of being destroyed. When addressing how to fix this issue, we initially considered fixing individual crashes we observed. However, the changes required to fix those problems would introduce considerable overhead to the nominal case. This is not reasonable in order to prevent a crash from occurring while Asterisk is already shutting down. Instead, this change makes it so AMI and CLI commands cannot be executed if Asterisk is being shut down. For AMI, this is absolute. For CLI, though, certain commands can be registered so that they may be run during Asterisk shutdown. ASTERISK-25825 #close Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990 2016-03-24 14:08 +0000 [3f720155b7] Alexander Traud * chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66 2016-03-19 07:34 +0000 [894071ea2c] Gianluca Merlo * config: fix flags in uint option handler The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e 2016-03-24 07:51 +0000 [13cdf3e8a1] Walter Doekes * musiconhold: Only warn if music class is not found in memory and database. The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf 2016-03-24 05:48 +0000 [87c9ab97ea] Walter Doekes * core/logging: Fix broken syslog levels on older glibc. The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However this macro is broken in older glibc (< 2.17); it would left-shift the facility a second time, causing the resultant priority to become invalid. The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: The priority argument is formed by ORing the facility and the level values [...]. ASTERISK-25510 #close Reported by: Michael Newton Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03 2016-03-24 06:18 +0000 [a72f3b5bb4] Joshua Colp * tests/test_http_media_cache: Fix file descriptor leak in test. Change-Id: Ie8a9ae3d13bdeaacafc8d28271adc6707f633a5f 2016-02-28 19:05 +0000 [13efea24f7] Matt Jordan * main/app: Only look to end of file if ':end' is specified, and not just ':' There is a little known feature in app_controlplayback that will cause the specified offset to be used relative to the end of a file if a ':end' is detected within the filename. This feature is pretty bad, but okay. However, a bug exists in this code where a ':' detected in the filename will cause the end pointer to be non-NULL, even if the full ':end' isn't specified. This causes us to treat an unspecified offset (0) as being "start playing from the end of the file", resulting in no file playback occurring. This patch fixes this bug by resetting the end pointer if ':end' is not found in the filename. Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35 2015-12-26 15:29 +0000 [ca14b99e6e] Matt Jordan * main/file: Add the ability to play media in the media cache This patch allows applications/APIs that access media through the core file APIs to play media in the media cache. Prior to determining if a 'filename' exists, the filename is passed to the media cache's retrieve API call. If that call succeeds, the local file specified passed back by the API is opened for streaming. When used in this fashion, the 'filename' is actually a URI that the media cache process and understand. ASTERISK-25654 #close Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0 2015-12-30 10:52 +0000 [01962a3932] Matt Jordan * tests/test_http_media_cache: Add unit tests for res_http_media_cache This patch adds unit tests for res_http_media cache, that covers nominal creation and retrieval - and through them as well, staleness and deletion checks. In addition, this patch adds tests that covers the interaction of various HTTP headers, including Expires, Etag, and Cache-Control. ASTERISK-25654 Change-Id: I2db101e307c863857fe416d6f5bf4cace9ac7cf5 2015-01-29 08:38 +0000 [22e2340813] Matt Jordan * res/res_http_media_cache: Add an HTTP(S) backend for the core media cache This patch adds a bucket backend for the core media cache that interfaces to a remote HTTP server. When a media item is requested in the cache, the cache will query its bucket backends to see if they can provide the media item. If that media item has a scheme of HTTP or HTTPS, this backend will be invoked. The backend provides callbacks for the following: * create - this will always retrieve the URI specified by the provided bucket_file, and store it in the file specified by the object. * retrieve - this will pull the URI specified and store it in a temporary file. It is then up to the media cache to move/rename this file if desired. * delete - destroys the file associated with the bucket_file. * stale - if the bucket_file has expired, based on received HTTP headers from the remote server, or if the ETag on the server no longer matches the ETag stored on the bucket_file, the resource is determined to be stale. Note that the backend respects the ETag, Expires, and Cache-Control headers provided by the HTTP server it is querying. ASTERISK-25654 Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250 2015-12-26 15:31 +0000 [791b4c9f81] Matt Jordan * main/media_cache: Provide an extension on the local file associated with a URI This patch does the following: First, it addresses file extension handling in the media cache. The media core in Asterisk is a bit interesting in that it wants: * A file to have an extension on it. That extension is used to associate the file with a defined format module. * The filename passed to the core to not have an extension on it. This allows the core to match the available file formats with the format a channel is capable of handling. Unfortunately, this makes the current implementation a bit lacking in the media cache. By default, we do not store the extension of a retrieved URI on the local file that is created. As a result, the media core does not know what format the file is, and the file is ignored. Modifying the file outside of the media core is bad, as we would not be able to update the internal ast_bucket_file's path. At the same time, we do not want to pass the extension out in the file_path parameter in ast_media_cache_retrieve. This parameter is intended to be fed into the media core; if we passed the extension, all callers would have to strip it off. Thus, this patch does the following: * If there is an extension specified in the URL, we append it to the local file name (if a preferred file name isn't specified), and we store that in the local file path. * The extension, however, is stripped off of the file_path parameter passed back out of ast_media_cache_retrieve. Second, this patch causes stale items to be completely removed from the system. Prior to this patch, sound files could be orphaned due to the bucket referencing the file being deleted, but the file itself not being removed. This is now addressed by explicitly calling ast_bucket_file_delete on the bucket_file when it is deemed to be stale. Note that this only happen when we know we will attempt to retrieve the resource again. Finally, this patch changes the AO2 container holding media items to just use a regular mutex. The usage for this container already assumed it was a plain mutex, and - given that retrieval of an item can cause it to be replaced in the container - a mutex makes more sense than a read/write lock. Change-Id: I51667fff86ae8d2e4a663555dfa85b11e935fe0f 2014-10-25 20:21 +0000 [6bbcfb34bd] Matt Jordan * funcs/func_curl: Add the ability for CURL to download and store files This patch adds a write option to the CURL dialplan function, allowing it to CURL files and store them locally. The value 'written' to the CURL URL specifies the location on disk to store the file. As an example: same => n,Set(CURL(http://1.1.1.1/foo.wav)=/tmp/foo.wav) Would retrieve the file foo.wav from the remote server and store it in the /tmp directory. Due to the potentially dangerous nature of this function call, APIs are forbidden from using the write functionality unless live_dangerously is set to True in asterisk.conf. ASTERISK-25652 #close Change-Id: I44f4ad823d7d20f04ceaad3698c5c7f653c41b0d 2016-03-23 08:59 +0000 [392341ba37] gtjoseph * pjproject-bundled: Cleanups for reported issues PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd 2015-11-20 08:02 +0000 [ac66999971] Francesco Castellano * chan_sip.c: Space after port causes unnecessary resolution attempt check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06 2016-03-19 17:49 +0000 [1d3191b118] gtjoseph * progdocs: Exclude ./third-party from documentation generation We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17 2016-03-18 20:32 +0000 [8f94f947f5] Gianluca Merlo * func_aes: fix misuse of strlen on binary data The encryption code for AES_ENCRYPT evaluates the length of the data to be encoded in base64 using strlen. The data is binary, thus the length of it can be underestimated at the first NULL character. Reuse the write pointer offset to evaluate it, instead. ASTERISK-25857 #close Change-Id: If686b5d570473eb926693c73461177b35b13b186 2016-03-18 14:31 +0000 [a3c9a74a02] Kevin Harwell * chan_pjsip: ref leak when checking direct_media_glare Fix the reference leak introduced in the following commit: c534bd58075e2e1a1e4f3b23c435186c71b155fd ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85 2016-03-16 12:37 +0000 [c534bd5807] Kevin Harwell * chan_pjsip: transfers with direct media reinvite has wrong address/port During a transfer involving direct media a race occurs between when the transferer channel is swapped out, initiating rtp changes/updates, and the subsequent reinvites. When Alice, after speaking with Charlie (Bob is on hold), connects Bob and Charlie invites are sent to each in order to establish the call between them. Bob is taken off hold and Charlie is told to have his media flow through Asterisk. However, if before those invites go out the bridge updates Bob's and/or Charlie's rtp information with direct media data (i.e. address, port) then the invite(s) will contain the remote data in the SDP instead of the Asterisk data. The race occurs in the native bridge glue code when updating the peer. The direct_media_address can get set twice before sending out the first invite during call connection. This can happen because the checking/setting of the direct_media_address happened in one thread while the sending of the invite(s) happened in another thread. This fix removes the race condition by moving the checking/setting of the direct_media_address to be in the same thread as the sending of the invites(s). This serializes the checking/setting and sending so they can no longer happen out of order. ASTERISK-25849 #close Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873 2016-03-03 04:43 +0000 [bdccb81157] Sergio Medina Toledo * res_pjsip_refer.c: Fix seg fault in process of Refer-to header. The "Refer-to" header of an incoming REFER request is parsed by pjsip_parse_uri(). That function requires the URI parameter to be NULL terminated. Unfortunately, the previous code added the NULL terminator by overwriting memory that may not be safe. The overwritten memory results could be benign, memory corruption, or a segmentation fault. Now the URI is NULL terminated safely by copying the URI to a new chunk of memory with the correct size to be NULL terminated. ASTERISK-25814 #close Change-Id: I32565496684a5a49c3278fce06474b8c94b37342 2016-02-25 10:29 +0000 [0da36fca6b] Leif Madsen * Add initial support to build Docker images This work-in-progress is the first step to being able to reliably build Asterisk containers from the Asterisk source. I'm submitting this based on feedback gained at AstriDevCon 2015. Information about how to use this is provided in contrib/docker/README.md and will result in a local Asterisk container being built right from your source. I believe this can eventually be automated via hub.docker.com. Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1 2016-03-11 12:22 +0000 [810f92c9dc] Richard Mudgett * chan_sip.c: Fix mwi resub deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 #close Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6 2016-03-10 17:01 +0000 [72c444ba37] Richard Mudgett * chan_sip.c: Fix registration timeout and expire deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508 2016-03-09 16:26 +0000 [7ea1e181dc] Richard Mudgett * chan_sip.c: Fix waitid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Made always run check_pendings() under the scheduler thread so scheduler ids can be checked safely. ASTERISK-25023 Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52 2016-03-10 12:17 +0000 [fbf8e04aed] Richard Mudgett * chan_sip.c: Fix t38id deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f 2016-03-08 15:08 +0000 [02458cc6fd] Richard Mudgett * chan_sip.c: Fix session timers deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900 2016-03-09 16:34 +0000 [c7fdff2e37] Richard Mudgett * chan_sip.c: Fix reinviteid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1 2016-03-07 13:21 +0000 [69810b306d] Richard Mudgett * chan_sip.c: Fix autokillid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f 2016-03-09 16:32 +0000 [f484ddbdfe] Richard Mudgett * chan_sip.c: Fix packet retransid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix retrans_pkt() to call check_pendings() with both the owner channel and the private objects locked as required. * Refactor dialog retransmission packet list to safely remove packet nodes. The list nodes are now ao2 objects. The list has a ref and the scheduled entry has a ref. ASTERISK-25023 Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641 2016-03-07 18:28 +0000 [67c79c326d] Richard Mudgett * chan_sip.c: Fix provisional_keepalive_sched_id deadlock. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48 2016-03-09 11:22 +0000 [76be7093cd] Richard Mudgett * chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. This patch is part of a series to resolve deadlocks in chan_sip.c. * Make dialog_unlink_all() unschedule all items at once in the sched thread. ASTERISK-25023 Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4 2016-03-10 21:54 +0000 [52f0932e4c] Richard Mudgett * chan_sip.c: Clear scheduled immediate events on unload. This patch is part of a series to resolve deadlocks in chan_sip.c. The reordering of chan_sip's shutdown is to handle any immediate events that get put onto the scheduler so resources aren't leaked. The typical immediate events at this time are going to be concerned with stopping other scheduled events. ASTERISK-25023 Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20 2016-03-15 14:51 +0000 [0987a11cce] Richard Mudgett * sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. This patch is part of a series to resolve deadlocks in chan_sip.c. Delaying destruction of the chan_sip sip_pvt structures caused the /channels/chan_sip/test_sip_rtpqos unit test to crash. That test registers a special test ast_rtp_engine with the rtp engine module. When the unit test completes it cleans up by unregistering the test ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt happens after the unit test returns, the destructor tries to call the rtp engine destroy callback of the test ast_rtp_engine auto variable which no longer exists on the stack. * Change the test ast_rtp_engine auto variable to a static variable. Now the variable can still exist after the unit test exits so the delayed sip_pvt destruction can complete successfully. ASTERISK-25023 Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13 2016-03-07 15:50 +0000 [9a7cfa2b61] Richard Mudgett * sched.c: Ensure oldest expiring entry runs first. This patch is part of a series to resolve deadlocks in chan_sip.c. * Updated sched unit test to check new behavior. ASTERISK-25023 Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3 2016-03-15 13:31 +0000 [7964e260d3] Andrew Nagy * app_stasis: Don't hang up if app is not registered This prevents pbx_core from hanging up the channel if the app isn't registered. ASTERISK-25846 #close Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce 2016-03-07 18:56 +0000 [cb97198ca6] Richard Mudgett * chan_sip.c: Simplify sip_pvt destructor call levels. Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a 2016-03-04 18:25 +0000 [8be01398d9] Richard Mudgett * chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12 2016-03-14 08:59 +0000 [4df7b3ae80] Joshua Colp * build: Add configure check for proto field of PJSIP TLS transport setting. Older versions of PJSIP do not have the proto field on the TLS transport setting structure. This change adds a configure check so even if it is not present we will still be able to build. Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9 2016-03-12 16:02 +0000 [0af6b5de62] gtjoseph * build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE I can't ever recall actually needing the intermediate files or the checking that a double compile produces. What I CAN remember is every DONT_OPTIMIZE build needing 3 invocations of gcc instead of 1 just to do the checks and produce those intermediate files. Having said that, Richard pointed out that the reason for the double compile was that there were cases in the past where a submitted patch failed to compile because the submitter never tried it with the optimizations turned on. To get the best of both worlds, COMPILE_DOUBLE has been split into its own option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected BUT you can then turn it off if all you need are the debugging symbols. This way you have to make an informed decision about disabling COMPILE_DOUBLE. To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature was added to menuselect. The element can now contain an "autoselect" attribute which will turn the used member on but not create a hard dependency. The cflags.xml implementation for COMPILE_DOUBLE looks like this... COMPILE_DOUBLE core * app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def 2016-03-07 21:34 +0000 [fb28049de2] gtjoseph * pjproject_bundled: Remove --with-external-pa from configure options. Not sure why it was there in the first place as we already specify --disable-sound. Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9 2016-03-06 14:38 +0000 [d2eb65f71e] gtjoseph * res_pjsip: Strip spaces from items parsed from comma-separated lists Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2 2016-03-07 02:02 +0000 [f690c105f3] Rodrigo Ramírez Norambuena * res_odbc_transaction: fix some format tab Change-Id: I265e4ac47c629c9a63dd86b59df82a7ab3c64384 2016-02-17 22:58 +0000 [0ec9fe5421] Rodrigo Ramírez Norambuena * main/cli.c: Refactor function to print seconds formatted Refactor and created function ast_cli_print_timestr_fromseconds to print seconds formatted: year(s) week(s) day(s) hour(s) second(s) This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c, res_config_ldap.c, res_config_pgsql.c. Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c 2016-03-04 20:37 +0000 [471ff375fd] gtjoseph * install_prereq: Add packages for bundled pjproject RedHat/CentOS needs python-devel Debian/Ubuntu needs automake, libsrtp-dev and python-dev Ubuntu also needed libncurses5-dev for cmenuselect so while not needed for pjproject, I adedd it anyway. Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089 2016-02-24 17:25 +0000 [2b9849625c] gtjoseph * res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited Per RFC3325, the 'From' header is now anonymized on outgoing calls when caller id presentation is prohibited. TID = trust_id_outbound PRO = Set(CALLERID(pres)=prohib) USR = endpoint/from_user DOM = endpoint/from_domain PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) Conditions |Result --------------------|---------------------------------------------------- TID PRO USR DOM |PAI FROM --------------------|---------------------------------------------------- Y Y abc def.ghi |PRI "Anonymous" Y Y abc |PRI "Anonymous" Y Y def.ghi |PRI "Anonymous" Y Y |PRI "Anonymous" Y N abc def.ghi |YES Y N abc |YES > Y N def.ghi |YES "Caller Name" @def.ghi> Y N |YES "Caller Name" @> N Y abc def.ghi |NO "Anonymous" N Y abc |NO "Anonymous" N Y def.ghi |NO "Anonymous" N Y |NO "Anonymous" N N abc def.ghi |YES N N abc |YES > N N def.ghi |YES "Caller Name" @def.ghi> N N |YES "Caller Name" @> ASTERISK-25791 #close Reported-by: Anthony Messina Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9 2016-03-03 17:34 +0000 [37472f7398] gtjoseph * third_party/Makefile.rules: Replace unsupported != operator with $(shell ...) Apparently the != operator is fairly new so I've replaced it with the old $(shell ...) syntax. Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479 Reported-by: Richard Mudgett 2016-01-23 15:50 +0000 [195100e770] gtjoseph * loader: Retry dlopen when loading fails Although we use the RTLD_LAZY flag when calling dlopen the first time on a module, this only defers resolution for function calls. Pointer references to functions are determined at link time so dlopen expects them to be there. Since we don't cross-module link, pointers to functions in other modules won't be available and dlopen will fail. Doing a "hardened" build also causes problems because it typically sets "-z now" on the ld command line which overrides RTLD_LAZY at run time. If the failing module isn't a GLOBAL_SYMBOLS module, then dlopen will be called again after all the GLOBAL_SYMBOLS modules have been loaded and they'll eventually resolve. If the calling module IS a GLOBAL_SYMBOLS module itself and a third module depends on it, then there's an issue because the second time through the dlopen loop, GLOBAL_SYMBOLS modules aren't given any special treatment and since the order in which dlopen is called isn't deterministic, the dependent may again be tried before the module it needs is loaded. Simple solution: Save modules that fail load_resource because of a dlopen error in a list and retry them immediately after the first pass. Keep retrying until the failed list is empty or we reach a #defined max retries. Error messages are suppressed until the final pass which also gets rid of those confusing error messages about module failures that are later corrected. Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb 2016-03-01 16:18 +0000 [15c5743ac1] Kevin Harwell * bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880 2016-03-01 18:08 +0000 [0d2ccbca62] Kevin Harwell * res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 During the transfer process, some phones (okay it was the Jitsi softphone, but maybe others are out there) send a "bye" immediately after receiving a SIP Notify. When a "bye" is received early for some types of transfers the transferer channel may no longer be available during late stage transfer processing. For instance, during an attended transfer involving stasis bridging at one point the created local channel looks for an associated swap channel in order to retrieve the stasis application name. If the transferer has hung up then the local channel will fail to find it. The local channel then has no way to know which stasis app to enter, so it fails and hangs up as well. Thus the transfer does not complete as expected. This patch delays the sending of the initial notify in order to give the transfer process enough time to gather the necessary data for a successful transfer. ASTERISK-25771 Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16 2016-03-03 08:26 +0000 [6af7fc4c37] Joshua Colp * res_pjsip_dtmf_info: NULL terminate the message body. PJSIP does not ensure that when printing the message body the buffer will be NULL terminated. This is problematic when searching for the signal and duration values of the DTMF. This change ensures the buffer is always NULL terminated. Change-Id: I52653a1a60c93092d06af31a27408d569cc98968 2016-03-01 20:03 +0000 [b8b7c2e428] gtjoseph * alembic: Fix downgrade and tweak for sqlite Downgrade had a few issues. First there was an errant 'update' statement in add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we weren't cleaning up the ENUMs so subsequent upgrades on postgres failed because the types already existed. For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. Fortunately alembic batch_operations takes care of this for us if we use it so the alter and drops were converted to use batch operations. Here's an example downgrade: with op.batch_alter_table('ps_endpoints') as batch_op: batch_op.drop_column('tos_audio') batch_op.drop_column('tos_video') batch_op.add_column(sa.Column('tos_audio', yesno_values)) batch_op.add_column(sa.Column('tos_video', yesno_values)) batch_op.drop_column('cos_audio') batch_op.drop_column('cos_video') batch_op.add_column(sa.Column('cos_audio', yesno_values)) batch_op.add_column(sa.Column('cos_video', yesno_values)) with op.batch_alter_table('ps_transports') as batch_op: batch_op.drop_column('tos') batch_op.add_column(sa.Column('tos', yesno_values)) # Can't cast integers to YESNO_VALUES, so dropping and adding is required batch_op.drop_column('cos') batch_op.add_column(sa.Column('cos', yesno_values)) Upgrades from base to head and downgrades from head to base were tested repeatedly for postgresql, mysql/mariadb, and sqlite3. Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8 2016-03-02 15:55 +0000 [7b71bca8a4] gtjoseph * config_transport: Fix objects returned by ast_sip_get_transport_states ast_sip_get_transport_states was returning a container of internal_state objects instead of ast_sip_transport_state objects. This was causing transport lookups to fail, most noticably in res_pjsip_nat, which couldn't find the correct external addresses. This was causing contacts to go out with internal ip addresses. ASTERISK-25830 #close Reported-by: Sean Bright Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e 2016-03-02 11:17 +0000 [0a3f0e85ac] Scott Griepentrog * CHAOS: cleanup possible null vars on msg alloc failure In message.c, if msg_alloc fails to init the string field, vars may be null, so use a null tolerant cleanup. In res_pjsip_messaging.c, if msg_data_create fails, mdata will be null, so use a null tolerant cleanup. ASTERISK-25323 Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56 2016-03-02 09:34 +0000 [60aa871be3] Scott Griepentrog * CHAOS: prevent crash on failed strdup This patch avoids crashing on a null pointer if the strdup() allocation fails. ASTERISK-25323 Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5 2016-02-29 18:11 +0000 [0bdbf0d882] Richard Mudgett * func_callerid.c: Update REDIRECTING reason documentation. Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386 2016-02-26 18:57 +0000 [25de01f301] Richard Mudgett * SIP diversion: Fix REDIRECTING(reason) value inconsistencies. Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87 2016-02-26 18:54 +0000 [8c8ef4efb0] Richard Mudgett * res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd 2016-02-29 20:41 +0000 [75ec137e91] Richard Mudgett * res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821 2016-01-18 21:54 +0000 [3173e91bab] gtjoseph * build-system: Allow building with static pjproject Background here: http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html From CHANGES: * To help insure that Asterisk is compiled and run with the same known version of pjproject, a new option (--with-pjproject-bundled) has been added to ./configure. When specified, the version of pjproject specified in third-party/versions.mak will be downloaded and configured. When you make Asterisk, the build process will also automatically build pjproject and Asterisk will be statically linked to it. Once a particular version of pjproject is configured and built, it won't be configured or built again unless you run a 'make distclean'. To facilitate testing, when 'make install' is run, the pjsua and pjsystest utilities and the pjproject python bindings will be installed in ASTDATADIR/third-party/pjproject. The default behavior remains building with the shared pjproject installation, if any. Building: All you have to do is include the --with-pjproject-bundled option on the ./configure command line (and remove any existing --with-pjproject option if specified). Everything else is automatic. Behind the scenes: The top-level Makefile was modified to include 'third-party' in the list of MOD_SUBDIRS. The third-party directory was created to contain any third party packages that may be needed in the future. Its Makefile automatically iterates over any subdirectories passing on targets. The third-party/pjproject directory was created to house the pjproject source distribution. Its Makefile contains targets to download, patch configure, generate dependencies, compile libs, apps and python bindings, sanitized build.mak and generate a symbols list. When bootstrap.sh is run, it automatically includes the configure.m4 file in third-party/pjproject. This file has a macro to download and conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR and PJPROJECT_BUNDLED. It also tests for the capabilities like PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to trying to compile. Of course, bootstrap.sh is only run once and the configure file is incldued in the patch. When configure is run with the new options, the macro in configure.m4 triggers the download, patch, conifgure and tests. No compilation is performed at this time. The downloaded tarball is cached in /tmp so it doesn't get downloaded again on a distclean. When make is run in the top-level Asterisk source directory, it will automatically descend all the subdirectories in third_party just as it does for addons, apps, etc. The top-level Makefile makes sure that the 'third-party' is built before 'main' so that dependencies from the other directories are built first. When main does build, a new shared library (libasteriskpj) is created that links statically to the pjproject .a files and exports all their symbols. The asterisk binary links to that, just as it does with libasteriskssl. When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject python bindings are installed in ASTDATADIR/third-party/pjproject. This will facilitate testing, including running the testsuite which will be updated to check that directory for the pjsua module ahead of the system python library. Modules should continue to depend on pjproject if they use pjproject APIs directly. They should not care about the implementation. No changes to any res_pjsip modules were made. Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103 2016-02-22 16:59 +0000 [2dae4a1ccf] Richard Mudgett * chan_sip.c: Fix T.38 issues caused by leaving a bridge. chan_sip could not handle AST_T38_TERMINATED frames being sent to it when the channel left the bridge. The action resulted in overlapping outgoing reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not happy. * Force T.38 to be remembered as locally bridged. Now when the channel leaves the native RTP bridge after T.38, the channel remembers that it has already reINVITEed the media back to Asterisk. It just needs to terminate T.38 when the AST_T38_TERMINATED arrives. * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if they happen before the T.38 state changes to disabled. Now the T.38 state is set to disabled before the reINVITE is sent. ASTERISK-25582 #close Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce 2016-02-18 18:27 +0000 [bf29a4e2e6] Richard Mudgett * res_pjsip_t38.c: Back out part of an earlier fix attempt. This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame that it puts into the bridge may or may not be processed by the time the bridged peer is kicked out of the bridge. If it is processed then all is well. However, if it is not processed then that channel is stuck in fax mode until it hangs up or maybe if it joins another bridge for T.38 faxing. ASTERISK-25582 Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7 2016-02-22 13:54 +0000 [c7d45b84f9] Richard Mudgett * bridge core: Add owed T.38 terminate when channel leaves a bridge. The channel is now going to get T.38 terminated when it leaves the bridging system and the bridged peers are going to get T.38 terminated as well. ASTERISK-25582 Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7 2016-02-19 16:01 +0000 [0e296563d7] Richard Mudgett * channel api: Create is_t38_active accessor functions. ASTERISK-25582 Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b 2016-02-19 19:06 +0000 [86f7336c91] Richard Mudgett * bridge_channel: Don't settle owed events on an optimization. Local channel optimization could cause DTMF digits to be duplicated. Pending DTMF end events would be posted to a bridge when the local channel optimizes out and is replaced by the channel further down the chain. When the real digit ends, the channel would get another DTMF end posted to the bridge. A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B 1) LocalA has the /n flag to prevent optimization. 2) B is sending DTMF to A through the local channel chain. 3) When LocalB optimizes out it can move B to the position of LocalB;1 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would settle an owed DTMF end to the bridge toward LocalA;2. 5) When B finally ends its DTMF it sends the DTMF end down the chain. 6) Without this patch, A would hear the DTMF digit end when LocalB optimizes out and when B ends the original digit. ASTERISK-25582 Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251 2016-02-22 12:15 +0000 [128c96456c] Richard Mudgett * channel.c: Route all control frames to a channel through the same code. Frame hooks can conceivably return a control frame in exchange for an audio frame inside ast_write(). Those returned control frames were not handled quite the same as if they were sent to ast_indicate(). Now it doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a channel or ast_indicate(). ASTERISK-25582 Change-Id: I5775f41421aca2b510128198e9b827bf9169629b 2016-02-25 15:13 +0000 [4422905218] gtjoseph * sorcery: Refactor create, update and delete to better deal with caches The ast_sorcery_create, update and delete function have been refactored to better deal with caches and errors. The action is now called on all non-caching wizards first. If ANY succeed, the action is called on all caching wizards and the observers are notified. This way we don't put something in the cache (or update or delete) before knowing the action was performed in at least 1 backend and we only call the observers once even if there were multiple writable backends. ast_sorcery_create was never adding to caches in the first place which was preventing contacts from getting added to a memory_cache when they were created. In turn this was causing memory_cache to emit errors if the contact was deleted before being retrieved (which would have populated the cache). ASTERISK-25811 #close Reported-by: Ross Beer Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46 2016-02-25 15:39 +0000 [acf329a3c7] gtjoseph * res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. There are a few cases where we're emitting notices or warnings for things that really need neither, like a client retrying to subscribe to mwi when they're not conifgured for it. They get a 404 so there's no need for non-debug messages. Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f 2016-02-25 14:17 +0000 [7e3e1ddf7e] gtjoseph * res_sorcery_memory_cache: Fix SEGV in some CLI commands A few of the CLI commands weren't checking for enough arguments and were SEGVing. Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413 2016-02-22 19:31 +0000 [803a2fc2d5] Richard Mudgett * rtp_engine.h: Remove extraneous semicolons. Change-Id: Ib462633d396fa941379dfef648dcd2245e350084 2016-02-23 14:57 +0000 [886ee09471] Richard Mudgett * chan_sip.c: Suppress T.38 SDP c= line if addr is the same. Use the correct comparison function since we only care if the address without the port is the same. Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0 2016-02-16 08:14 +0000 [b7970cabfa] Christof Lauber * res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables Introduced realloaction of ast_str buf in sqlite3_escape functions in case the returned buffer from threadstorage was actually too small. Change-Id: I3c5eb43aaade93ee457943daddc651781954c445 2016-02-11 11:01 +0000 [ba8adb4ce3] gtjoseph * res_pjsip/config_transport: Allow reloading transports. The 'reload' mechanism actually involves closing the underlying socket and calling the appropriate udp, tcp or tls start functions again. Only outbound_registration, pubsub and session needed work to reset the transport before sending requests to insure that the pjsip transport didn't get pulled out from under them. In my testing, no calls were dropped when a transport was changed for any of the 3 transport types even if ip addresses or ports were changed. To be on the safe side however, a new transport option was added (allow_reload) which defaults to 'no'. Unless it's explicitly set to 'yes' for a transport, changes to that transport will be ignored on a reload of res_pjsip. This should preserve the current behavior. Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf 2016-02-19 04:30 +0000 [c00082329e] Walter Doekes * chan_sip: Optionally supply fromuser/fromdomain in SIP dial string. Previously you could add [!dnid] to the SIP dial string to alter the To: header. This change allows you to alter the From header as well. SIP dial string extra options now look like this: [![touser[@todomain]][![fromuser][@fromdomain]]] INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To: header, that is no longer possible. ASTERISK-25803 #close Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7 2016-02-07 17:34 +0000 [f8767a8804] gtjoseph * res_pjproject: Add ability to map pjproject log levels to Asterisk log levels Warnings and errors in the pjproject libraries are generally handled by Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings or errors so the messages emitted by pjproject directly are either superfluous or misleading. A good exampe of this are the level-0 errors pjproject emits when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS client be treated any differently? A config file for res_pjproject has bene added (pjproject.conf) and a new log_mappings object allows mapping pjproject levels to Asterisk levels (or nothing). The defaults if no pjproject.conf file is found are the same as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898 2016-02-18 10:55 +0000 [14886643c6] Alexei Gradinari * res_pjsip_outbound_publish: Fix processing 412 response When Asterisk receives a 412 (Conditional Request Failed) response it has to recreate publish session. There is bug in res_pjsip_outbound_publish.c The function sip_outbound_publish_client_alloc is called with wrong object while processing 412 (Conditional Request Failed) response. This patch fixes it. ASTERISK-25229 #close Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359 2016-02-18 11:15 +0000 [8055d080cd] Mark Michelson * Fix failing threadpool_auto_increment test. The threadpool_auto_increment test fails infrequently for a couple of reasons * The threadpool listener was notified of fewer tasks being pushed than were actually pushed * The "was_empty" flag was set to an unexpected value. The problem is that the test pushes three tasks into the threadpool. Test expects the threadpool to essentially gather those three tasks, and then distribute those to the threadpool threads. It also expects that as the tasks are pushed in, the threadpool listener is alerted immediately that the tasks have been pushed. In reality, a task can be distributed to the threadpool threads quicker than expected, meaning that the threadpool has already emptied by the time each subsequent task is pushed. In addition, the internal threadpool queue can be delayed so that the threadpool listener is not alerted that a task has been pushed even after the task has been executed. From the test's point of view, there's no way to be able to predict exactly the order that task execution/listener notifications will occur, and there is no way to know which listener notifications will indicate that the threadpool was previously empty. For this reason, the test has been updated to only check the things it can check. It ensures that all tasks get executed, that the threads go idle after the tasks are executed, and that the listener is told the proper number of tasks that were pushed. Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c 2016-02-17 13:30 +0000 [30a49b8a6a] Richard Mudgett * cel.c: Fix mismatch in ast_cel_track_event() return type. The return type of ast_cel_track_event() is not large enough to return all 64 potential bits of the event enable mask. Fortunately, the defined CEL events do not really need all 64 bits and the return value is only used to determine if the requested CEL event is enabled. * Made the ast_cel_track_event() return 0 or 1 only so the return value can fit inside an int type instead of zero or a truncated 64 bit non-zero value. Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c 2016-02-16 23:37 +0000 [15aeb78c66] Rodrigo Ramírez Norambuena * app_queue: fix Calculate talktime when is first call answered Fix calculate of average time for talktime is wrong when is completed the first call beacuse the time for talked would be that call. ASTERISK-25800 #close Change-Id: I94f79028935913cd9174b090b52bb300b91b9492 2016-02-16 16:37 +0000 [62282bb8ce] gtjoseph * res_odbc: Fix exports.in for missing symbols res_odbc.exports.in was missing a few symbols. Changed to wildcards. Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c 2016-02-16 12:20 +0000 [49203628f9] gtjoseph * res_statsd: Fix exports.in for missing symbols res_statsd.export.in was missing the _va variations of the log functions causing Asterisk to crash in res_pjsip if OPTIONAL_API wasn't enabled. ASTERISK-25727 #close Reported-by: Gergely Dömsödi Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b 2016-02-15 21:31 +0000 [4f08e9fb64] gtjoseph * res_pjsip_config_wizard: Add command to export primitive objects A new command (pjsip export config_wizard primitives) has been added that will export all the pjsip objects it created to the console or a file suitable for reuse in a pjsip.conf file. ASTERISK-24919 #close Reported-by: Ray Crumrine Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b 2016-02-15 15:37 +0000 [be811c4be1] gtjoseph * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify the header added by the dialplan function. Since the header added by the dialplan function is generic string, there are no virtual functions to parse the uri and we get a segfault when we try. Since the modify, was really only an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER and recreate it. This raises a question for another time though: What should happen with duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups so if it's session supplement is loaded after res_pjsip_caller_id's (or any other module that adds headers), there'll be dups in the message. ASTERISK-25337 #close Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa 2016-02-15 13:08 +0000 [13b6c02945] Mark Michelson * Fix creation race of contact_status structures. It is possible when processing a SIP REGISTER request to have two threads end up creating contact_status structures in sorcery. contact_status is created using a "find or create" function. If two threads call into this at the same time, each thread will fail to find an existing contact_status, and so both will end up creating a new contact status. During testing, we would see sporadic failures because the PJSIP_CONTACT() dialplan function would operate on a different contact_status than what had been updated by res_pjsip/pjsip_options. The fix here is two-fold: 1) The "find or create" function for contact_status now has a lock around the entire operation. This way, if two threads attempt the operation simultaneously, the first to get there will create the object, and the second will find the object created by the first thread. 2) res_sorcery_memory has had its create callback updated so that it will not allow for objects with duplicate IDs to be created. Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97 2016-02-15 12:52 +0000 [5c400a0fed] Joshua Colp * res_pjsip_pubsub: Move where the subscription is stored to after initialized. A problem arose when testing the AMI subscription listing actions where it was possible for a subscription that had not been fully initialized to be listed. This was problematic as the underlying listing code would crash. This change makes it so the subscription tree is fully set up before it is added to the list of subscriptions. This ensures that when the listing actions get the subscription it is valid. ASTERISK-25738 #close Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48 2016-02-09 17:34 +0000 [b37555cc94] gtjoseph * res_pjsip: Refactor load_module/unload_module load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 2016-02-09 22:42 +0000 [c4d9f46878] Badalyan Vyacheslav * Resources/res_phoneprov: fix memory leak and heap-use-after-free * heap-use-after-free happens when we free "cfg" but then use "value" which refers to it * A memory leak occurs because in some cases it is not released "defaults" ASTERISK-25721 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469 2016-02-11 11:21 +0000 [e5fd972d24] Etienne Lessard (license #6394) * func_iconv: Ensure output strings are properly terminated. ASTERISK-25272 #close Reported by: Etienne Lessard patches: AST-25272.patch submitted by Etienne Lessard (license #6394) Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17 2016-02-10 16:16 +0000 [168c18737f] gtjoseph * res_pjsip: Handle pjsip_dlg_create_uas deprecation Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically increments the lock on the returned dialog. To account for this, configure.ac now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use the original call or the new one. If the new one was used, the ref count is decremented before returning. ASTERISK-25751 #close Reported-by Josh Colp Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8 2016-02-09 20:13 +0000 [fd668670b5] Rodrigo Ramírez Norambuena * res_config_pgsql: Show error message in reload if not connected. Change-Id: I9290115a1aaadb589eb1d02eaeb502eec01b31fa 2016-02-09 23:40 +0000 [a23d01e943] Badalyan Vyacheslav * Build: Added testing compiler to support the system sanitizes In older versions of the compiler was not sanitizes. Compilers other than GCC can not support the Usan and TSAN or have other options for *FLAGS. ASTERISK-25767 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916 2016-02-09 20:57 +0000 [c7186c7f0a] Badalyan Vyacheslav * Build: Fix menuselect USAN conflicts USAN can be used together with other sanitizers. Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f 2016-02-09 14:21 +0000 [68643f83cd] Corey Farrell * Simplify and fix conditional in FD_SET. FD_SET contains a conditional statement to protect against buffer overruns. The statement was overly complicated and prevented use of the last array element of ast_fdset. We now just verify the fd is less than ast_FDMAX. Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40 2016-02-09 07:11 +0000 [e40fddbeb5] Joshua Colp * tests/test_sorcery_memory_cache_thrash: Improve termination process. When terminating the threads thrashing a sorcery memory cache each would be told to stop and then we would wait on them. During at least one thrashing test this was problematic due to the specific usage pattern in use. It would take some time for termination of the thread to occur. This would occur due to contention between the threads retrieving and the threads updating the cache. As the retrieving threads are given priority it may be some time before the updating threads are able to proceed. This change makes it so all threads are told to stop and then each are joined to ensure they stop. This way all the threads should stop at around the same time instead of waiting for one to stop, the next to stop, then the next, and so on. As a result of this the execution time for each thrash test is much closer to their expected value than previously seen as well. Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827 2016-01-29 17:56 +0000 [bbf3ace682] gtjoseph * res_pjsip: Fix infinite recursion when loading transports from realtime Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19 2016-02-07 13:00 +0000 [72bf53eea5] Rodrigo Ramírez Norambuena * res_config_pgsql: Add message on cli failed command status In case failed of command "realtime show pgsql status" show a message the data of connection to more clear information in error. Change-Id: Ia8e9e2400466606e7118f52a46e05df0719b6a29 2016-02-05 10:29 +0000 [b69729dde5] gtjoseph * chan_misdn: Fix a few issues causing compile errors Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98 2016-01-25 17:36 +0000 [1bc54aee80] Richard Mudgett * app_confbridge: Only use b_profile options from the conference. A user cannot set new bridge options after the conference is created by the first user. Attempting to do so is documented as undefined behavior. This patch ensures that the bridge profile options used are from the conference and not what a subsequent user may have tried to set. Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266 2016-02-04 16:17 +0000 [3b426a8b09] Mark Michelson * Check for OpenSSL defines before trying to use them. The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior to OpenSSL version 1.0.1. A recent commit attempts to, by default, set these options, which can cause problems on systems with older OpenSSL installations. This commit adds a configure script check for those defines and will not attempt to make use of those if they do not exist. We will print a warning urging the user to upgrade their OpenSSL installation if those defines are not present. Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d 2016-02-03 14:25 +0000 [9b13ab6a63] gtjoseph * pjsip/alembic: Add missing columns to system and registration ps_systems needed disable_tcp_switch ps_registrations needed line and endpoint ASTERISK-25737 #close Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19 2016-02-04 11:39 +0000 [82e2938fa8] Mark Michelson * res_stasis_device_state: Fix refcounting error. Device state subscription lifetimes were governed by when the subscription was established and unsubscribed from. However, it is possible that at the time of unsubscription, there could be device state events still in flight. When those device state events occur, the device state callback could attempt to dereference a freed pointer. Crash. This change ensures that the lifetime of the device state subscription does not end until the underlying stasis subscription has confirmed that its final message has been sent. Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2 2016-01-27 10:44 +0000 [d83dba7099] Sean Bright * res_rtp_asterisk: Allow ICE host candidates to be overriden During ICE negotiation the IPs of the local interfaces are sent to the remote peer as host candidates. In many cases Asterisk is behind a static one-to-one NAT, so these host addresses will be internal IP addresses. To help in hiding the topology of the internal network, this patch adds the ability to override the host candidates by matching them against a user-defined list of replacements. Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f 2016-02-03 12:05 +0000 [0de74fad55] Joshua Colp * AST-2016-001 http: Provide greater control of TLS and set modern defaults. This change exposes the configuration of various aspects of the TLS support and sets the default to the modern standards. The TLS cipher is now set to the best values according to the Mozilla OpSec team, different TLS versions can now be disabled, and the cipher order can be forced to be that of the server instead of the client. ASTERISK-24972 #close Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8 2015-12-07 12:46 +0000 [e67b445e8d] Richard Mudgett * AST-2016-003 udptl.c: Fix uninitialized values. Sending UDPTL packets to Asterisk with the right amount of missing sequence numbers and enough redundant 0-length IFP packets, can make Asterisk crash. ASTERISK-25603 #close Reported by: Walter Doekes ASTERISK-25742 #close Reported by: Torrey Searle Change-Id: I97df8375041be986f3f266ac1946a538023a5255 2015-09-28 17:07 +0000 [a877e0d94b] Richard Mudgett * AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. Setting the sip.conf timert1 value to a value higher than 1245 can cause an integer overflow and result in large retransmit timeout times. These large timeout times hold system file descriptors hostage and can cause the system to run out of file descriptors. NOTE: The default sip.conf timert1 value is 500 which does not expose the vulnerability. * The overflow is now detected and the previous timeout time is calculated. ASTERISK-25397 #close Reported by: Alexander Traud Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290 2016-02-03 14:07 +0000 [dcbedf9ab1] gtjoseph * logging: Remove/fix some message annoyances test_dlinklists doesn't need to NOTICE everyone that every macro worked. res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or provider was registered. res_odbc was missing a newline at the end of one message. Change-Id: I6c06361518ef3711821795e535acd439782a995e 2016-02-02 10:52 +0000 [6522361871] Alexei Gradinari License #5691 * res_sorcery_realtime: Fix regex regression. A regression was introduced where searching for realtime PJSIP objects by regex by starting the regex with a leading "^" would cause no items to be returned. This was due to a change which attempted to drop the requirement for a leading "^" to be present due to how some CLI commands formulate their regexes. However, the change, rather than simply eliminating the requirement, caused any regexes that did begin with "^" to end up not returning the expected results. This change fixes the problem by inspecting the regex and formulating the realtime query differently depending on if it begins with "^". ASTERISK-25702 #close Reported by Nic Colledge Patches: realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693 2016-02-02 04:05 +0000 [2a6f18cd55] Karsten Wemheuer * res_xmpp: Does not connect in component mode The module res_xmpp does not accept usernames in the form used in component mode (XEP-0114). In component mode there is no @something in the name. In component mode the connection is now not dropped anymore. If the xmpp server sends out a "stream" tag before handshake is finished, the connection gets dropped in res_xmpp. Now this tag will be ignored and the connection will be established. After connecting there will be an exchange of presence states. This does not work as expected in component mode. The responsible function "xmpp_pak_presence" is left before the states get sent out. Sending presence states in component mode is now moved to the top of the function. ASTERISK-25735 #close Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c 2016-02-01 13:04 +0000 [40da6434c1] gtjoseph * build_system: Fix some warnings highlighted by clang Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd 2016-01-31 20:13 +0000 [52b29f9b4c] gtjoseph * pjsip/alembic: Fix definition of qualify_timeout A recent commit set qualify_timeout to Decimal which isn't supported. This path corrects it to Float. Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf 2016-01-29 07:39 +0000 [55a7367ad4] Stefan Engström * chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) AMI action: SIPnotify or b) cli command: sip notify , I expect asterisk to include the same value for its own ip in both cases a) and b), but it seems a) produces a contact header like Contact: whereas b) produces a contact header like . 0.0.0.0:8060 is my udpbindaddr in sip.conf My guess is that manager_sipnotify should call ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, because after applying this patch, both cases a) and b) produce the contact header that I expect: Reported by: Stefan Engström Tested by: Stefan Engström Change-Id: I86af5e209db64aab82c25417de6c768fb645f476 2016-01-28 12:44 +0000 [d2397f028f] Richard Mudgett * config_options.c: Fix warning message wording. Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4 2016-01-25 17:34 +0000 [af6b15976d] Richard Mudgett * app_confbridge.c: Replace inlined code with existing function. Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51 2016-01-25 16:05 +0000 [7932336a3d] Richard Mudgett * app_confbridge: Add ability to get the muted conference state. * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. * Added Muted header to AMI ConfbridgeListRooms action response list events to indicate the muted conference state. * Added Muted column to CLI "confbridge list" output to indicate the muted conference state and made the locked column a yes/no value instead of a locked/unlocked value. ASTERISK-20987 Reported by: hristo Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1 2016-01-26 17:59 +0000 [894045e7cf] Richard Mudgett * app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7 2016-01-25 15:48 +0000 [12c93e8f81] Richard Mudgett * app_confbridge: Make non-admin users join a muted conference muted. ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38 2016-01-27 13:08 +0000 [f19bf7a321] gtjoseph * res_pjsip: Add res_pjproject dependency to samples Since res_pjsip now depends on res_pjproject, this has been added to basic-pbx modules.conf. Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d 2016-01-27 10:29 +0000 [c53903d447] gtjoseph * build_system: Prevent goals needing makeopts from running when it's missing The Makefile only optionally includes makeopts so when goals like uninstall that dont depend on anything else are run after a distclean, rules like 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts to remove everything in the root directory. Although there's a rule defined for makeopts which prints a message and does an 'exit 1', since '-include makepopts' was specified (with the -), the exit was ignored letting the rest of the rules run. This patch makes makeopts required unless the goal has the string 'clean' in it. ASTERISK-25730 #close Reported-by: George Joseph Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7 2016-01-25 09:35 +0000 [1dfd104a27] Joshua Colp * config: Allow options to register when documentation is unavailable. The config options framework is strict in that configuration options must be documented unless XML documentation support is not available. In practice this is useful as it ensures documentation exists however in off-nominal cases this can cause strange problems. If it is expected that a config option has a non-zero or non-empty default value but the config option documentation is unavailable this reasonable expectation will not be met. This can cause obscure crashes and weirdness depending on how the code handles it. This change tweaks the behavior to ensure that the config option is still allowed to register, apply default values, and be set when devmode is not enabled. If devmode is enabled then the option can NOT be set. This also does not remove the initial documentation error message that is output on load when registering the configuration option. ASTERISK-25725 #close Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8 2016-01-25 10:23 +0000 [a706ad44e6] Mark Michelson * Stasis: Use custom structure when setting variables. A recent change to queue channel variable setting to the Stasis control queue caused a regression. When setting channel variables, it is possible to give a NULL channel variable value in order to unset the variable (i.e. remove it from the channel variable list). The change introduced a call to ast_variable_new(), which is not tolerant of NULL channel variable values. This new change switches from using ast_variable to using a custom channel variable struct that is lighter weight and NULL value-tolerant. Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d 2016-01-25 16:56 +0000 [289daca9e8] Rusty Newton * sounds/Makefile: Incremented core and extra sounds versions to 1.5 Core and extra sounds 1.5 was recently released! The tarballs contain change descriptions however I figure more people will see this one so I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra to Core for en, en_GB, fr and added for languages that didn't already have Extra sound sets (it,ja,ru). In addition all of the English and Russian sounds have been completely re-recorded. Sounds moved and added: activated,added,all-circuits-busy-now,astcc-followed-by-pound at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy ,call-fwd-unconditional,calling,call-waiting,cancelled, cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to ,location,number,number-not-answering,num-was-successfully,one-moment-please ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered ,your There were also a few random fixes here and there to file names for a few of the languages. ASTERISK-25068 #close Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3 2016-01-25 16:51 +0000 [b073244c51] Mark Michelson * res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. A test recently uncovered that running an ill-timed AMI command to show inbound subscriptions could cause a crash since Asterisk will try to operate on a freed subscription. The fix for this is to remove the subscription tree from the list of subscriptions at the time that we are sending our final NOTIFY request out. This way, as the subscription is in the process of dying, it is inaccessible from AMI. Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 2016-01-25 11:03 +0000 [830f8933c2] Corey Farrell * chan_sip: Fix buffer overrun in sip_sipredirect. sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer of 256 characters. This patch reduces the copy to 255 characters to leave room for the string null terminator. ASTERISK-25722 #close Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab 2016-01-23 16:45 +0000 [f299dc0d76] Rodrigo Ramírez Norambuena * app_queue: Add Lastpause field of queue member Add time when started a the last pause for a queue member for QueueMemberStatus ami event. Also show accumulate time in seconds when started a pause for a queue member to CLI command 'queue show'. ASTERISK-16394 #close Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c 2016-01-23 12:34 +0000 [8c664da0ff] Rodrigo Ramírez Norambuena * app_queue: fix some tab format Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c 2016-01-23 11:41 +0000 [2fb45c7801] Rodrigo Ramírez Norambuena * cdr_pgsql.cl: REFACTOR Macro LENGTHEN_BUF Remove repeated code on macro of assigned buffer to SQL vars. Add table and connection name to log error message when is not possible allocate memory. Change-Id: I1fbf37d286a032d38fdda72a9f736356956c9ffe 2016-01-22 15:08 +0000 [959f7436cc] Mark Michelson * Stasis: Fix potential memory leak of control data. When queuing tasks onto the Stasis control queue, you can pass an arbitrary data pointer and a function to free that data. All ARI commands that use the Stasis control queue made the assumption that the destructor function would be called in all paths, whether the task was queued successfully or not. However, this was not correct. If a task was queued onto a control structure that was already completed, the allocated data would not be freed properly. This patch corrects this by making sure that all return paths call the data destructor. Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb 2016-01-21 10:58 +0000 [a45eacebf3] Mark Michelson * Stasis: Use control queue to prevent crash. A crash occurred when attempting to set a channel variable on a channel that had already been hung up. This is because there is a small window between when a control is grabbed and when the channel variable is set that the channel can be hung up. The fix here is to queue the setting of the channel variable onto the control queue. This way, the manipulation of the channel happens in a thread where it is safe to be done. In this change, I also noticed that the setting of bridge roles on channels was being done outside of the control queue, so I also changed those operations to be done in the control queue. ASTERISK-25709 #close Reported by Mark Michelson Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78 2016-01-22 11:48 +0000 [7866806fc3] Richard Mudgett * logger.c: Fix buffer overrun found by address sanitizer. The null terminator of the tail struct member was not being allocated when no logger.conf config file is installed. ASTERISK-25714 #close Reported by: Badalian Vyacheslav Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30 2015-12-23 15:07 +0000 [9714da7aa4] Mark Michelson * res_odbc: Remove connection management Asterisk by default will create a single database connection and share it among all threads that attempt to access the database. In previous versions of Asterisk, this was tolerable, because the most used channel driver, chan_sip, mostly accessed the database from a single thread. With PJSIP, however, many threads may be attempting to perform database operations, and there is the potential for many more database accesses, meaning the concurrency is a horrible bottleneck if only one connection is shared. Asterisk has a connection pooling facility built into it, but the implementation has flaws. For one, there is a strict limit on the number of simultaneous connections that could be made to the database. Anything beyond the maximum would result in a failed operation. Attempting to predict what the maximum should be is nearly impossible even for someone intimately familiar with Asterisk's threading model. In addition, use of transactions in the dialplan can cause some severe bugs if connection pooling is enabled. This commit seeks to fix the concurrency problem by removing all connection management code from Asterisk and leaving that to the underlying unixODBC code instead. Now, Asterisk does not share a single connection, nor does it try to maintain a connection pool. Instead, all Asterisk ever does is request a connection from unixODBC and allow unixODBC to either allocate those connections or retrieve them from a pool. Doing this has a bit of a ripple effect. For one, since connections are not long-lived objects, several of the safeguards that previously existed have been removed. We don't have to worry about trying to use a connection that has gone stale. In every case, when we request a connection, it has just been made and we don't need to perform any sanity checks to be sure it's still active. Another major player affected by this change is transactions. Transactions and their respective connections were so tightly coupled that it was almost pornographic. This code change moves transaction-related code to its own file separate from the core ODBC functionality. This way, the core of ODBC does not even have to know that transactions exist. In making this large change, I had to look at a lot of code and understand it. When making this change, I discovered several places where the behavior is definitely not ideal, but it seemed outside the scope of this change to be fixing it. Instead, any place where I saw some sort of room for improvement has had a XXX comment added explaining what could be altered to improve it. Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf 2016-01-22 11:18 +0000 [d3969d09ae] Rodrigo Ramírez Norambuena * app_queue.c: remove include for core_unreal.h not used in code. Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d 2016-01-21 16:40 +0000 [5dde111719] Corey Farrell * Build System: Add support for checking alembic branches. * Add 'check-alembic' target to root Makefile. * Create build_tools/make_check_alembic to do the actual checks. ASTERISK-25685 Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004 2016-01-19 18:20 +0000 [04078f43b5] Richard Mudgett * res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. ASTERISK-25712 #close Reported by: Richard Mudgett Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f 2016-01-13 16:49 +0000 [5615db3714] Richard Mudgett * res_pjsip: Add CLI "pjsip dump endpt [details]" Dump the res_pjsip endpt internals. In non-developer mode we will not document or make easily accessible the "details" option even though it is still available. The user has to know it exists to use it. Presumably they would also be aware of the potential crash warning below. Warning: PJPROJECT documents that the function used by this CLI command may cause a crash when asking for details because it tries to access all active memory pools. Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb 2016-01-18 03:49 +0000 [b259ac95ac] Diederik de Groot * main/asterisk.c: ast_el_read_char Make sure buf[res] is not accessed at res=-1 (buffer underrun). Address Sanitizer will complain about this quite loudly. ASTERISK-24801 #close Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9 2016-01-18 19:27 +0000 [dd5c063934] gtjoseph * res_pjproject: Add module providing pjproject logging and utils res_pjsip_log_forwarder has been renamed to res_pjproject and enhanced as follows: As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, a new ast_pjproject_get_buildopt function has been added. It allows the caller to get the value of one of the buildopts. The initial use case is retrieving the runtime value of PJ_MAX_HOSTNAME to insure we don't send a hostname greater than pjproject can handle. Since it can differ between the version of pjproject that Asterisk was compiled against and the version of pjproject that Asterisk is running against, we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk source code. Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e 2016-01-18 17:16 +0000 [3b9cba4294] Matt Jordan * funcs/func_cdr: Correctly report high precision values for duration and billsec When CDRs were refactored, func_cdr's ability to report high precision values for duration and billsec (the 'f' option) was broken. This was due to func_cdr incorrectly interpreting the duration/billsec values provided by the CDR engine in milliseconds, as opposed to seconds. Since the CDR engine only provides duration and billsec in seconds, and does not expose either attribute with sufficient precision to merely pass back the underlying value, this patch fixes the bug by re-calculating duration and billsec with microsecond precision based on the start/answer/end times on the CDR. ASTERISK-25179 #close Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841 2016-01-20 07:52 +0000 [479cc99acd] Rodrigo Ramírez Norambuena * README: Update year in copyright Change-Id: I56240f537fb3205672cdb2a74f0591ae7bb73dbc 2016-01-19 17:15 +0000 [9fa76ba215] Joshua Colp * test_threadpool: Wait for each task to complete and fix memory leak. This change makes the thread_timeout_thrash unit test wait for each task to complete. This fixes the problem where the test would prematurely end when all threads were gone and a new one had to be started to handle the last task. It also increases the thrasing as it is now more likely for each task to encounter the above scenario. This also fixes a memory leak where the data for each task was not being freed. ASTERISK-25611 #close Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6 2016-01-18 19:44 +0000 [c9f7269b2e] Richard Mudgett * taskprocessor.c: Increase CLI "core ping taskprocessor" timeout. Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029 2016-01-18 19:43 +0000 [6e2a867716] Richard Mudgett * taskprocessor.c: Fix some taskprocessor unrefs. You have to call ast_taskprocessor_unref() outside of the taskprocessor implementation code. Taskprocessor use since v12 has become more transient than just the singleton uses in earlier versions. Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb 2016-01-19 14:16 +0000 [a4dcbdf50f] Richard Mudgett * Fix alembic branches on master. Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8 2016-01-05 17:12 +0000 [35a3e8cc7f] Corey Farrell * Refactor init_logger_chain locking. This removes logchannels locking from init_logger_chain, puts the responsibility on the caller. Adds locking around the one call that was missing it. ASTERISK-24833 Change-Id: I6cc42117338bf9575650a67bcb78ab1a33d7bad8 2016-01-18 22:10 +0000 [378fed4900] Rodrigo Ramírez Norambuena * app_queue: Fix preserved reason of pause when Asterisk is restared When the Asterisk is restared is not preseved reason paused of members. This patch fixed this cases, retain data on astdb and set when Asterisk is started. ASTERISK-25732 #close Report by: Rodrigo Ramírez Norambuena Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5 2016-01-18 19:01 +0000 [130aa1427e] gtjoseph * pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b 2016-01-16 13:18 +0000 [eaf2b5052e] Daniel Journo * Update version number in features.conf.sample Update the version number in the comments from Asterisk 12 to Asterisk 12+ Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b 2016-01-13 15:58 +0000 [c60d6c0162] Daniel Journo * pjsip/alembic: Fix qualify_timeout column definition Corrects the qualify_timeout column type from Integer to Decimal ASTERISK-25686 #close Reported-by: Marcelo Terres Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8 2016-01-15 19:52 +0000 [480ccfcc97] Corey Farrell * main/config: Clean config maps on shutdown. ASTERISK-25700 #close Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808 2016-01-14 14:42 +0000 [a5b38b604c] Kevin Harwell * bridge_basic: don't cache xferfailsound during an attended transfer The xferfailsound was read from the channel at the beginning of the transfer, and that value is "cached" for the duration of the transfer. Therefore, changing the xferfailsound on the channel using the FEATURE() dialplan function does nothing once the transfer is under way. This makes it so the transfer code instead gets the xferfailsound configuration options from the channel when it is actually going to be used. This patch also fixes a potential memory leak of the props object as well as making sure the condition variable gets initialized before being destroyed. ASTERISK-25696 #close Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4 2015-07-10 10:37 +0000 [d36c4d0b01] Richard Mudgett * taskprocessor.c: Simplify ast_taskprocessor_get() return code. Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1 2016-01-13 18:20 +0000 [0a878020dc] Richard Mudgett * astmm.c: Add more stats to CLI "memory show" commands. * Add freed regions totals to allocations and summary. * Add totals for all allocations and not just the selected allocations. Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a 2016-01-14 16:00 +0000 [84b30c5e18] Kevin Harwell * bridge_basic: don't play an attended transfer fail sound after target hangs up If the attended transfer destination answers (picks call up or goes to voicemail) and then hangs up on the transferer then transferer hears the fail sound. This patch makes it so the fail sound is not played when the transfer destination/target hangs up after answering. ASTERISK-25697 #close Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded 2016-01-14 14:36 +0000 [c7caee6c4b] Corey Farrell * Remove *.gcna / *.gcno files from added module sources. Asterisk uses a Makefile macro to associate additional sources with a module. This macro is responsible for creating clean targets but previously left behind *.gcna and *.gcno files. ASTERISK-25683 #close Reported by yaron nahum Change-Id: Idc0823fe80a25c42cefae901fde875e9fc38d8ea 2016-01-14 09:26 +0000 [68cad96ffd] Rusty Newton * func_channel: Add help text for undocumented CHANNEL function arguments Adding help text documentation for: * hangupsource * appname * appdata * exten * context * channame * uniqueid * linkedid ASTERISK-24097 #close Reported by: Steven T. Wheeler Tested by: Rusty Newton Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d 2016-01-10 16:22 +0000 [8182146e85] Daniel Journo * pjsip: Add option global/regcontext Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62 2016-01-12 11:14 +0000 [022423b98b] Joshua Colp * app: Queue hangup if channel is hung up during sub or macro execution. This issue was exposed when executing a connected line subroutine. When connected or redirected subroutines or macros are executed it is expected that the underlying applications and logic invoked are fast and do not consume frames. In practice this constraint is not enforced and if not adhered to will cause channels to continue when they shouldn't. This is because each caller of the connected or redirected logic does not check whether the channel has been hung up on return. As a result the the hung up channel continues. This change makes it so when the API to execute a subroutine or macro is invoked the channel is checked to determine if it has hung up. If it has then a hangup is queued again so the caller will see it and stop. ASTERISK-25690 #close Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea 2016-01-13 07:20 +0000 [79a7321a47] Sean Bright * res_musiconhold: Prevent multiple simultaneous reloads. There are two ways in which the reload() function in res_musiconhold can be called from the CLI: * module reload res_musiconhold.so * moh reload In the former case, the module loader holds a lock that prevents multiple concurrent calls, but in the latter there is no such protection. This patch changes the 'moh reload' CLI command to invoke the module loader directly, rather than call reload() explicitly. ASTERISK-25687 #close Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c 2016-01-12 14:25 +0000 [1fffe71f77] Richard Mudgett * res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". PJPROJECT has a function available to dump the compile time options used when building the library. * Add CLI "pjsip show buildopts" command. * Update contrib/scripts/autosupport to get pjproject information. Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748 2016-01-12 10:36 +0000 [01c5e2a07e] Mark Michelson * res_sorcery_realtime: Remove leading ^ requirement. res_sorcery_realtime's search-by-regex callback performed a check to ensure that the passed-in regex began with a caret (^). If it did not, then no results would be returned. This callback only started to become used when "like" support was added to PJSIP CLI commands. The CLI command for listing objects would pass an empty regex ("") to the sorcery backend if no "like" statement was present. For most sorcery backends, this resulted in returning all objects. However, for realtime, this resulted in returning no objects. This commit seeks to fix the regression by removing the requirement from res_sorcery_realtime for the passed-in-regex to begin with a caret. ASTERISK-25689 #close Reported by Marcelo Terres Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20 2016-01-07 11:57 +0000 [a41aab477a] gtjoseph * pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88 2016-01-08 16:59 +0000 [7760029f19] Kevin Harwell * pbx: Deadlock between contexts container and context_merge locks Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5) introduced the possibility of a deadlock. Due to the mentioned modifications ast_change_hints now needs to keep both merge/delete and state callbacks from occurring while it executes. Unfortunately, sometimes ast_change_hints can be called with the contexts container locked. When this happens it's possible for another thread to grab the context_merge_lock before the thread calling into ast_change_hints does and then try to obtain the contexts container lock. This of course causes a deadlock between the two threads. The thread calling into ast_change_hints waits for the other thread to release context_merge_lock and the other thread is waiting on that one to release the contexts container lock. Unfortunately, there is not a great way to fix this problem. When hints change, the subsequent state callbacks cannot run at the same time as a merge/delete, nor when the usual state callbacks do. This patch alleviates the problem by having those particular callbacks (the ones run after a hint change) occur in a serialized task. By moving the context_merge_lock to a task it can now safely be attempted or held without a deadlock occurring. ASTERISK-25640 #close Reported by: Krzysztof Trempala Change-Id: If2210ea241afd1585dc2594c16faff84579bf302 2016-01-10 17:08 +0000 [e9c2c1dc67] Corey Farrell * devicestate: Cleanup engine thread during graceful shutdown. ASTERISK-25681 #close Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1 2016-01-10 13:51 +0000 [90c0dcaee4] Corey Farrell * manager: Cleanup manager_channelvars during shutdown. ASTERISK-25680 #close Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446 2016-01-10 13:27 +0000 [a868a381f0] Corey Farrell * res_calendar: Cleanup scheduler context at unload. ASTERISK-25679 #close Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f 2016-01-08 11:49 +0000 [a1c43022d2] Joshua Colp * res_rtp_asterisk: Revert DTLS negotiation changes. Due to locking issues within pjnath these changes are being reverted until pjnath can be changed. ASTERISK-25645 Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays." This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d. Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705 Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation" This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b. Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe 2016-01-09 17:57 +0000 [220ba979cf] gtjoseph * Revert "pjsip_location: Delete contact_status object when contact is deleted" This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45. Matt, This patch causes another problem and should not have been needed. Before this patch, persistent_endpoint_contact_deleted_observer WAS deleting the contact_status when ast_sip_location_delete_contact was called. By deleting it yourself in ast_sip_location_delete_contact it was gone before the observer could run and the observer therefore was throwing an error and not sending stasis/AMI/statsd messages. So, I don't think this was the cause of your original issue. I also had verified the contact AMI and statsd lifecycle and it was working. I'll double check now though. ASTERISK-25675 Reported-by: Daniel Journo Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a 2016-01-09 18:04 +0000 [26e0e113dc] Corey Farrell * pbx_dundi: Run cleanup on failed load. During failed startup of pbx_dundi no cleanup was performed. Add a call to unload_module before returning AST_MODULE_LOAD_DECLINE. ASTERISK-25677 #close Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29 2016-01-09 13:28 +0000 [dc2c000fd5] Corey Farrell * res_crypto: Perform cleanup at shutdown. This change causes res_crypto to unregister CLI at shutdown while still preventing the module from being unloaded. ASTERISK-25673 #close Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc 2016-01-06 19:10 +0000 [0bca2a5c26] Richard Mudgett * res_pjsip: Create human friendly serializer names. PJSIP name formats: pjsip/aor/- -- registrar thread pool serializer pjsip/default- -- default thread pool serializer pjsip/messaging -- messaging thread pool serializer pjsip/outreg/- -- outbound registration thread pool serializer pjsip/pubsub/- -- pubsub thread pool serializer pjsip/refer/- -- REFER thread pool serializer pjsip/session/- -- session thread pool serializer pjsip/websocket- -- websocket thread pool serializer Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084 2016-01-06 19:09 +0000 [f0f5fbbc01] Richard Mudgett * Sorcery: Create human friendly serializer names. Sorcery name formats: sorcery/- -- Sorcery thread pool serializer Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47 2016-01-06 19:09 +0000 [b1c7ae9afc] Richard Mudgett * Stasis: Create human friendly taskprocessor/serializer names. Stasis name formats: subm:- -- Stasis subscription mailbox task processor subp:- -- Stasis subscription thread pool serializer Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd 2016-01-07 16:15 +0000 [3e857bb347] Richard Mudgett * taskprocessor.c: New API for human friendly taskprocessor names. * Add new API call to get a sequence number for use in human friendly taskprocessor names. * Add new API call to create a taskprocessor name in a given buffer and append a sequence number. Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9 2016-01-06 17:19 +0000 [84c245d38c] Richard Mudgett * taskprocessor.c: Fix CLI "core show taskprocessors" output format. Update the CLI "core show taskprocessors" output format to not be distorted because UUID names are longer than previously used taskprocessor names. Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601 2016-01-07 21:07 +0000 [7d86979ea0] Richard Mudgett * taskprocessor.c: Fix CLI "core show taskprocessors" unref. Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5 2016-01-06 19:00 +0000 [1fb39aa8a0] Richard Mudgett * ccss.c: Replace space in taskprocessor name. The CLI "core ping taskprocessor" command does not work very well with taskprocessor names that have spaces in them. You have to put quotes around the name so using tab completion becomes awkward. Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0 2016-01-07 20:44 +0000 [71bb7b9c40] Richard Mudgett * taskprocessor.c: Sort CLI "core show taskprocessors" output. Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e 2016-01-05 16:54 +0000 [b025e1982f] Richard Mudgett * taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b 2015-12-16 11:25 +0000 [c5e16fe33a] Mark Michelson * Alembic: Add PJSIP global keep_alive_interval. The keep_alive_interval option was added about a year ago, but no alembic revision was created to add the appropriate column to the database. This commit fixes the problem and adds the column. This was discovered by running the testsuite with automatic conversion to realtime enabled. Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a 2016-01-07 03:21 +0000 [6745cd6529] Diederik de Groot * include/asterisk/time.h: Renamed global declaration:tv Renamed global declaration:tv to dummy_tv_var_for_types, which would oltherwise cause 'shadow' warnings when 'tv' was declared as a local variable elsewhere. Added comment to note that dummy_tv_var_for_types is never really exported and only used as a place holder. ASTERISK-25627 #close Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28 2016-01-07 15:37 +0000 [1afc8432dc] Mark Michelson * PJSIP: Prevent deadlock due to dialog/transaction lock inversion. A deadlock was observed where the monitor thread was stuck, therefore resulting in no incoming SIP traffic being processed. The problem occurred when two 200 OK responses arrived in response to a terminating NOTIFY request sent from Asterisk. The first 200 OK was dispatched to a threadpool worker, who locked the corresponding transaction. The second 200 OK arrived, resulting in the monitor thread locking the dialog. At this point, the two threads are at odds, because the monitor thread attempts to lock the transaction, and the threadpool thread loops attempting to try to lock the dialog. In this case, the fix is to not have the monitor thread attempt to hold both the dialog and transaction locks at the same time. Instead, we release the dialog lock before attempting to lock the transaction. There have also been some debug messages added to the process in an attempt to make it more clear what is going on in the process. ASTERISK-25668 #close Reported by Mark Michelson Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a 2016-01-07 09:39 +0000 [5d8c42c6d3] Corey Farrell * ast_format_cap_append_by_type: Resolve codec reference leak. This resolves a reference leak caused by ASTERISK-25535. The pointer returned by ast_format_get_codec is saved so it can be released. ASTERISK-25664 #close Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec 2016-01-07 03:33 +0000 [7856762f2f] Diederik de Groot * main: Use ast_strdup instead of strdup Fix compile error in main/utils.c because strdup was used in dummy_start Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793 2016-01-06 07:12 +0000 [64b2046f3d] Walter Doekes * Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. The spandspflow2pcap.py creates pcap files from fax.log files, generated through 'fax set debug on' when receiving a fax. An example fax.log is included as spandspflow2pcap.log. The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a recent version of SIPp. ASTERISK-25660 #close Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526 2016-01-04 04:26 +0000 [084563e136] Aaron An * cel/cel_radius: Fix wrong pointer. The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter y not the address of y. I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, then i review the source code and compare it with cdr/cdr_radius.c. Fix it and it works. ASTERISK-25647 #close Reported by: Aaron An Tested by: Aaron An Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0 2016-01-04 20:23 +0000 [36f1eaf0b5] Corey Farrell * main/pbx: Move hangup handler routines to pbx_hangup_handler.c. This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves hangup handler management functions to their own source. Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104 2015-12-21 11:07 +0000 [90b06d1a3c] Martin Tomec * app_queue: Add member flag "in_call" to prevent reading wrong lastcall time Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 2016-01-04 19:46 +0000 [3507494b8a] Corey Farrell * main/pbx: Move dialplan application management routines to pbx_app.c. This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves dialplan application management functions to their own source. Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c 2016-01-04 18:20 +0000 [54a8f1a396] Corey Farrell * main/pbx: Move switch routines to pbx_switch.c. This is the fifth patch in a series meant to reduce the bulk of pbx.c. This moves ast_switch functions to their own source. Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e 2016-01-04 18:00 +0000 [c3c8b8e41d] Corey Farrell * main/pbx: Move timing routines to pbx_timing.c. This is the fourth patch in a series meant to reduce the bulk of pbx.c. This moves pbx timing functions to their own source. Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6 2015-12-30 10:49 +0000 [6d18fe151c] gtjoseph * voicemail: Move app_voicemail / res_mwi_external conflict to runtime The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 2016-01-04 16:15 +0000 [5ee5c3739e] Corey Farrell * main/pbx: Move variable routines to pbx_variables.c. This is the third patch in a series meant to reduce the bulk of pbx.c. This moves channel and global variable routines to their own source. Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6 2015-12-04 17:22 +0000 [f88b952093] Richard Mudgett * app_dial: Immediately exit dial if the caller is already hung up. If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 2016-01-02 10:26 +0000 [e9dd16364e] Matt Jordan * main/cdr: Allow setting properties on a finalized CDR if it is the last one Prior to this patch, we explicitly disallowed setting any properties on a finalized CDR. This seemed like a good idea at the time; in practice, it was more restrictive. There are weird and strange scenarios where setting a property on a finalized CDR is definitely wrong. For example, we may Fork a CDR, finalizing the previous one, then change a property. In said case, the old CDR is supposed to now be 'immutable' (so to speak), and should not be updated. From the perspective of the code, a forked CDR that is finalized is just finalized. Hence why we decided these should not be updated. In practice, it is much more common to want to set a property on a CDR in the h extension or in a hangup handler. Disallowing a common scenario to make an esoteric behaviour work isn't good. This patch fixes this by allowing callers to set a property IF we are the last CDR in the chain. This preserves the finalized CDR if it was forked, while allowing the more common case to function. ASTERISK-25458 #close Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9 2016-01-02 10:23 +0000 [153547a9b1] Matt Jordan * main/cdr: Set the end time on a CDR if endbeforehexten is Yes Prior to this patch, the CDR engine attempted to set the end time on a CDR that was executing hangup logic and with endbeforehexten set to Yes by calling a function that inspects the properties on the Party A snapshot to determine if we are ready to set the end time. That always failed. This is because a Party A snapshot is not updated for CDRs that are executing hangup logic with endbeforehexten=Yes. Instead of calling a function that looks at the Party A snapshot, we just simply set the end time on the CDR. This is safe to call multiple times, and is safe to call at this point as we know that (a) we are executing hangup logic, and (b) we are supposed to set the end time at this point. ASTERISK-25458 Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3 2015-12-30 20:51 +0000 [f9bfc2450e] Corey Farrell * main/pbx: Move custom function routines to pbx_functions.c. This is the second patch in a series meant to reduce the bulk of pbx.c. This moves custom function management routines to their own source. Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177 2016-01-01 05:25 +0000 [3fd528dddf] Rodrigo Ramírez Norambuena * Happy new year 2016. Change-Id: I22d3c90f6f27df82e915bbf81c1d91221f7a945e 2015-12-13 13:09 +0000 [9cdf3ec19d] Matt Jordan * res_pjsip_history: Add a module that provides PJSIP history for debugging This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36 2015-12-28 19:18 +0000 [5e67e51c6a] gtjoseph * main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c We joked about splitting pbx.c into multiple files but this first step was fairly easy. All of the pbx_builtin dialplan applications have been moved into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() is called by asterisk.c just after load_pbx(). A few functions were renamed and are cross-exposed between the 2 source files. Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a 2015-12-28 14:02 +0000 [a05bb258b1] Joshua Colp * test_time: Provide a timeout when waiting. The test_timezone_watch unit test is written to expect a condition to be signaled when the inotify daemon thread runs. There exists a small window where the test_timezone_watch thread can signal the inotify daemon thread while it is not reading on the underlying file descriptor. If this occurs the test_timezone_watch thread will wait indefinitely for a signal that will never arrive. This change adds a timeout to the condition so it will return regardless after a period of time. Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 2015-12-24 20:26 +0000 [96b32e0321] Matt Jordan * tests/test_stasis_endpoints: Remove expected duplicate events The cache_clear test was written to expect duplicate Stasis messages sent from the technology endpoint to the all caching topic. This patch fixes the test to no longer expect these duplicate messages. ASTERISK-25137 Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981 2015-12-24 22:19 +0000 [3bddcc0219] Dade Brandon * res_http_websocket.c: prevent avoidable disconnections caused by write errors Updated ast_websocket_write to encode the entire frame in to one write operation, to ensure that we don't end up with a situation where the websocket header has been sent, while the body can not be written. Previous to August's patch in commit b9bd3c14, certain network conditions could cause the header to be written, and then the sub-sequent body to fail - which would cause the next successful write to contain a new header, and a new body (resulting in the peer receiving two headers - the second of which would be read as part of the body for the first header). This was patched to have both write operations individually fail by closing the websocket. In a case available to the submitter of this patch, the same body which would consistently fail to write, would succeed if written at the same time as the header. This update merges the two operations in to one, adds debug messages indicating the reason for a websocket connection being closed during a write operation, and clarifies some variable names for code legibility. Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598 2015-05-27 13:22 +0000 [22db16fa81] gtjoseph * endpoint/stasis: Eliminate duplicate events on endpoint status change When an endpoint is created, its messages are forwarded to both the tech endpoint topic and the all endpoints topic. This is done so that various parties interested in endpoint messages can subscribe to just the tech endpoint and receive all messages associated with that particular technology, as opposed to subscribing to the all endpoints topic. Unfortunately, when the tech endpoint is created, it also forwards all of its messages to the all topic. This results in duplicate messages whenever an endpoint publishes its messages. This patch resolves the duplicate message issue by creating a new function for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts as a normal caching topic, save that it no longer forwards messages it receives to the all endpoints topic. This allows it to act as an aggregation "sink", while preserving the necessary caching behaviour. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b 2015-12-27 22:38 +0000 [6b08f01c60] Corey Farrell * Remove res_jabber file that was left behind. Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b 2015-12-26 09:24 +0000 [d4b10cfb3e] Ward van Wanrooij * chan_sip: option 'notifyringing' change and doc fix In the sample sip.conf this is written with regard to notifyringing: ;notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) However, this setting changes whether or not any RINGING indications are sent to subscriptions. There is no separate configurable setting that allows to control whether INUSE subscriptions also get sent RINGING. This is however a useful option, to see (using BLF) if somebody else is able to handle an incoming call or if everybody is busy. This patch corrects the documentation for notifyringing (so the documentation matches the functionality) and make notifyringing a tri-state option, by adding the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing = notinuse, only subscriptions that are not INUSE are sent the RINGING signal. The default setting for notifyringing remains set to yes, so the default behaviour is not affected. ASTERISK-25558 Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa 2015-12-25 09:56 +0000 [6dc21bbf00] Dade Brandon * chan_sip.c: fix websocket_write_timeout default value websocket_write_timeout was not being set to its default value during sip config reload, which meant that prior to this commit, 1) the default value of 100 was not used, unless an invalid value (or 1) was specified in sip.conf for websocket_write_timeout, and 2) if the websocket_write_timeout directive was removed from sip.conf without a full restart of asterisk, then the previous value would continue to be used indefinitely. This essentially lead to a 0ms write timeout (the first write attempt in ast_careful_fwrite must have succeeded) in websocket write requests from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. Changes to websocket_write_timeout still only apply to new websocket sessions, after the sip reload -- timeouts on existing sessions are not adjusted during sip reload. Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953 2015-12-23 17:40 +0000 [8eb5da0679] Richard Mudgett * bridge_basic.c: Fix GOTO_ON_BLINDXFR Use of GOTO_ON_BLINDXFR would not work at all. The target location would never be executed by the transferring channel. * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with valid context, exten, and priority parameters from the transferring channel. * Renamed some feature_blind_transfer() local variables for clarity. ASTERISK-25641 #close Reported by Dmitry Melekhov Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac 2015-12-24 12:19 +0000 [2df4ad647c] Matt Jordan * res/res_pjsip_location: Delete contact_status object when contact is deleted In 450579e908, a change was made that removed the deletion of the 'contact_status' object when a 'contact' object is deleted in sorcery. This unfortunately means that the 'contact_status' object persists, even when something has explicitly removed a contact. The result is that the state of the contact will not be regenerated if that contact is re-created, and the stale state will be reported/used for that contact. It also results in no ContactStatusChanged events being generated for either ARI or AMI. This patch restores the deletion logic that was removed. Doing so now results in the expected events being generated again. Change-Id: I28789a112e845072308b5b34522690e3faf58f07 2015-12-24 10:18 +0000 [b8876711f3] Kevin Harwell * res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151 2015-12-20 21:33 +0000 [ca394161cf] Dade Brandon * app_amd: Correct maximum_number_of_words functionality & documentation - The maximum_number_of_words was previously documented as being the number of words that when exceeded, would result in the AMD application returning that the audio represents a machine. This was inconsistent with its actual functionality - it was a number of words that when REACHED, would result in determination as a machine. This update corrects the functionality to match the previously documented functionality. This is a backwards incompatible change in configuration file, and has been added to UPGRADE.txt as a result. The sample configuration file and application defaults have been updated so that the default value is now 2, which reflects the same default functionality as previous versions. - Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. - Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093 2015-12-17 19:05 +0000 [648ca2b1b8] Dade Brandon * res_rtp_asterisk: Resolve further timing issues with DTLS negotiation Resolves an edge case dtls negotiation delay for certain networks which somehow manage to drop the rtcp side's packet when these are both sent ast_rtp_remote_address_set, causing it to have to time-out and restart the handshake. Move dtls pending bio flush in to it's own function, and call it from ast_rtp_on_ice_complete, when we're rtp->ice, rather than when ast_rtp_remote_address_set. Keep the existing flush from the recent change to res_rtp_remote_address_set if ice is not being used. ASTERISK-25614 #close Reported-by: XenCALL Tested by: XenCALL Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5 2015-12-05 10:01 +0000 [902309fd04] Joshua Colp * res_sorcery_memory_cache: Add support for a full backend cache. This change introduces the configuration option 'full_backend_cache' which changes the cache to be a full mirror of the backend instead of a per-object cache. This allows all sorcery retrieval operations to be carried out against it and is useful for object types which are used in a "retrieve all" or "retrieve some" pattern. ASTERISK-25625 #close Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5 2015-12-17 10:25 +0000 [a2431f83ef] Joshua Colp * rtp_engine: Ignore empty filenames in DTLS configuration. When applying an empty DTLS configuration the filenames in the configuration will be empty. This is actually valid to do and each filename should simply be ignored. Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539 2015-12-17 08:10 +0000 [d2c8614122] Joshua Colp * chan_sip: Enable WebSocket support by default. Per the documentation the WebSocket support in chan_sip is supposed to be enabled by default but is not. This change corrects that. Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423 2015-12-14 12:04 +0000 [d17d9a9288] Joshua Colp * json: Audit ast_json_* usage for thread safety. The JSON library Asterisk uses, jansson, is not thread safe for us in a few ways. To help with this wrappers for JSON object reference count increasing and decreasing were added which use a global lock to ensure they don't clobber over each other. This does not extend to reference count manipulation within the jansson library itself. This means you can't safely use the object borrowing specifier (O) in ast_json_pack and you can't share JSON instances between objects. This change removes uses of the O specifier and replaces them with the o specifier and an explicit ast_json_ref. Some cases of instance sharing have also been removed. ASTERISK-25601 #close Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1 2015-12-16 11:28 +0000 [cfb34adb83] Mark Michelson * Alembic: Increase column size of PJSIP AOR "contact". When running the PJSIP AMI "show_endpoint" test with automatic conversion to realtime, the test would fail. This was because the AOR "contact" column was sized at 40, and the configured contact was larger than that. This commit increases the size of the contact column to 255 characters. Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1 2015-12-14 13:53 +0000 [32ec83f37f] server-pandora * res_rtp_asterisk.c: Fix DTLS negotiation delays. - Trigger pending DTLS packets to send out, once the RTP instance's remote address is set. - Avoids locking the DTLS structure unnecessarily by only doing this if DTLS is passive. - Add DTLS locks around the structurally sensitive calls in the SSL portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock inside of itself, and we're dealing with the SSL BIO in at least two threads. WebRTC channels may receive a DTLS handshake before ast_rtp_remote_address_set is called, which causes there to be a pending response to send out. Previous to 1ad827, this was handled by calling dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP packet could trigger the pending handshake response. Since that was rightfully removed, whenever the DTLS handshake is received before the remote address is set, we would have to wait until another SSL packet arrives. As of Chrome M47's optimizations to their handshake process, WebRTC conversations between Chrome M47+ and Asterisk, where Asterisk is passive, experience a 1 second delay without this patch, because the SSL handshake is received before ICE negotation stores the remote_address, and the next SSL packet isn't received until after a 1 second timeout in Chrome, which causes a new handshake request. ASTERISK-25614 #close Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908 2015-12-08 13:04 +0000 [52ca6fb94a] sungtae kim * AMI: Fixed OriginateResponse message When the asterisk sending OriginateResponse message, it doesn't set the "Uniqueid". And it didn't support correct response message for Application originate. ASTERISK-25624 #close Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d 2015-12-14 15:25 +0000 [eccdf2250b] Richard Mudgett * Fix sscanf() format string type mismatch. ASTERISK-25615 Reported by: George Joseph Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b 2015-12-14 06:26 +0000 [3e7522533c] Carlos Oliva * app_queue: update RT members when the 1st call joins a queue with no agents If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682 2015-12-13 13:13 +0000 [9a96a86e2d] Matt Jordan * main/utils: Don't emit an ERROR message if the read end of a pipe closes An ERROR or WARNING message should generally indicate that something has gone wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not in control of when the far end closes its reading on a file descriptor. If the far end does close the file descriptor in an unclean fashion, this isn't a bug or error in Asterisk, particularly when the situation can be gracefully handled in Asterisk. Currently, when this happens, a user would see the following somewhat cryptic ERROR message: "utils.c: write() returned error: Broken pipe" There's a few problems with this: (1) It doesn't provide any context, other than 'something broke a pipe' (2) As noted, it isn't actually an error in Asterisk (3) It can get rather spammy if the thing breaking the pipe occurs often, such as a FastAGI server (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even mask legitimate issues This patch changes ast_carefulwrite to only log an ERROR if we actually had one that was reasonably under our control. For debugging purposes, we still emit a debug message if we detect that the far side has stopped reading. Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566 2015-12-12 11:08 +0000 [3e6637feb5] gtjoseph * pjsip/config_transport: Check pjproject version at runtime for async ops pjproject < 2.5.0 will segfault on a tls transport if async_operations is greater than 1. A runtime version check has been added to throw an error if the version is < 2.5.0 and async_operations > 1. To assist in the check, a new api "ast_compare_versions" was added to utils which compares 2 major.minor.patch.extra version strings. ASTERISK-25615 #close Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 Reported-by: George Joseph Tested-by: George Joseph 2015-12-10 11:44 +0000 [ceebdfce40] Jonathan Rose * chan_sip: Add TCP/TLS keepalive to TCP/TLS server Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36 2015-12-07 13:07 +0000 [fcaebb0e43] Corey Farrell * app_meetme: Set default value for audio_buffers. The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44 2015-12-09 09:48 +0000 [5790700497] Tyler Cambron * res_chan_stats: Fix bug to send correct statistics to StatsD Fixed a bug that originally would show a negative number of active calls occuring in Asterisk. A gauge is persistent so incrementing and decrementing it results in a more consistent performance. Also changed to the call to StatsD to use ast_statsd_log_string() so that a "+" could be sent to StatsD. ASTERISK-25619 #close Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7 2015-12-08 17:49 +0000 [a987434564] gtjoseph * res_pjsip: Add existence and readablity checks for tls related files Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph 2015-12-02 12:42 +0000 [be693539c3] Eugene Voityuk * chan_sip.c: Start ICE negotiation when response is sent or received. The current logic for ICE negotiation starts it when receiving an SDP with ICE candidates. This is incorrect as ICE negotiation can only start when each call party have at least one pair of local and remote candidate. Starting ICE negotiation early would result in negotiation failure and ultimately no audio. This change makes it so ICE negotiation is only started when a response with SDP is received or when a response with SDP is sent. ASTERISK-24146 Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca 2015-12-08 01:57 +0000 [59a91c350a] Filip Jenicek * chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) Asterisk may crash when calling ast_channel_get_t38_state(c) on a locked channel which is being hung up. ASTERISK-25609 #close Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b 2015-12-08 11:03 +0000 [28ab03fbf7] gtjoseph * res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls See ASTERISK-25615. If the transport protocol is tls and async_operations > 1, pjproject will segfault if more than one operation is attempted on the same socket. Until this is fixed upstream, a check has been added to throw an error if a tls transport config has async_operations set to > 1. ASTERISK-25615 Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6 Reported-by: George Joseph Tested-by: George Joseph 2015-12-08 08:39 +0000 [55dd7125b3] Alexander Traud * codec_resample: Increase buffer for Opus Codec with FEC. ASTERISK-25599 #close Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e 2015-12-08 03:46 +0000 [64f899e5f3] Alexander Traud * translate: Avoid a warning message when doing FEC within Opus Codec. ASTERISK-25616 #close Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319 2015-12-04 15:36 +0000 [65c8147952] Richard Mudgett * chan_sip: Fix crash involving the bogus peer during sip reload. A crash happens sometimes when performing a CLI "sip reload". The bogus peer gets refreshed while it is in use by a new call which can cause the crash. * Protected the global bogus peer object with an ao2 global object container. ASTERISK-25610 #close Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed 2015-11-13 07:58 +0000 [48c065e46d] Christof Lauber * chan_sip: Support parsing of Q.850 reason header in SIP BYE and CANCEL requests. Current support for reason header did work only in SIP responses. According to RFC3336 the reason header might appear in any SIP request. But it seems to make most sence in BYE and CANCEL so parasing is done there too (if use_q850_reason=yes). Change-Id: Ib6be7b34c23a76d0e98dfd0816c89931000ac790 2015-12-06 16:35 +0000 [75c800eb28] Matt Jordan * Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" This reverts commit f42d22d3a1ca5c8ea73df99a50c6a28caa8f8749. Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks in core_unreal/chan_local. Local channels attempt to reach across both their peer and the peer's bridge to inspect T.38 state. Given the propensity of Local channel chains, managing the locking situation in such a scenario is practically infeasible. Change-Id: I932107387c13aad2c75a7a4c1e94197a9d6d8a51 2015-12-04 16:23 +0000 [4be231e82f] gtjoseph * res_pjsip/contacts/statsd: Make contact lifecycle events more consistent It will never be perfect or even pretty, mostly because of the differences between static and dynamic contacts. Created: Can't use the contact or contact_status alloc functions because the objects come and go regardless of the actual state. Can't use the contact_apply_handler, ast_sip_location_add_contact or a sorcery created handler because they only get called for dynamic contacts. Similarly, permanent_uri_handler only gets called for static contacts. So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is the only place it can go and not have duplicated code. Both permanent_uri_handler and contact_apply_handler call find_or_create. Removed: Can't use the destructors for the same reason as above. The only place to put this is in persistent_endpoint_contact_deleted_observer which I believe is the "correct" place but even that will handle only dynamic contacts. This doesn't called on shutdown however. There is no hook to use for static contacts that may be removed because of a config change while asterisk is in operation. I moved the cleanup of contact_status from ast_sip_location_delete_contact to the handler as well. Status Change and RTT: Although they worked fine where they were (in update_contact_status) I moved them to persistent_endpoint_contact_status_observer to make it more consistent with removed. There was logic there already to detect a state change. Finally, fixed a nit in permanent_uri_handler rmudgett reported eralier. ASTERISK-25608 #close Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d Reported-by: George Joseph Tested-by: George Joseph 2015-11-21 06:08 +0000 [63c6d39a3e] Alexander Traud * res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. ASTERISK-25584 #close Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91 2015-11-28 08:46 +0000 [f42d22d3a1] Matt Jordan * bridges/bridge_t38: Add a bridging module for managing T.38 state When 4875e5ac32 was merged, it fixed several issues with a direct media bridge transitioning to handling a T.38 fax. However, it uncovered a race condition caused by the bridging core. When a channel involved in a T.38 fax leaves a bridge, the frame queued by the channel driver that should inform the far side that it is no longer in a T.38 fax may not make it across the bridge. The bridging framework is *extremely* aggressive in tearing down the bridge, and control frames that are currently in flight *may* get dropped. This patch adds a new module to the bridging framework, bridge_t38. This module maintains some notion of the T.38 state for the two channels in a bridge. When the bridge detects that it is being torn down or when one of the two channels leaves, it informs the respective channel(s) that they should stop faxing. This ensures that channels switch back to audio if they survive and are ejected out of a bridge while faxing. ASTERISK-25582 Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0 2015-11-21 05:35 +0000 [dcc01bc0a7] Alexander Traud * res_format_attr_opus: Update to latest RFC 7587. Beside that, the format-attribute module sends only non-default values in the line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, previously the parameter stereo was not parsed when being the first parameter. ASTERISK-25583 #close Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73 2015-12-02 14:11 +0000 [69457b8d61] Jonathan Rose * Fix crash in audiohook translate to slin This patch fixes a crash which would occur when an audiohook was applied to a channel using an audio codec that could not be translated to signed linear (such as when using pass-through codecs like OPUS or when the codec translator module for the format in use is not loaded). ASTERISK-25498 #close Reported by: Ben Langfeld Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384 2015-12-03 12:07 +0000 [5959186017] gtjoseph * res_pjsip: Use a MD5 hash for static Contact IDs When 90d9a70789 was merged, it mostly tested dynamic contacts created as a result of registering a PJSIP endpoint. Contacts generated in this fashion typically have a long alphanumeric string as their object identifier, which maps reasonably well for StatsD. Unfortunately, this doesn't work in the general case. StatsD treats both '.' and ':' characters as special characters. In particular, having a ':' appear in the middle of a StatsD metric will result in the metric being rejected. This causes some obvious issues with SIP URIs. The StatsD API should not be responsible for escaping the metric name passed to it. The metric is treated as a single long string, and it would be challenging to know what to escape in the string passed to the function. Likewise, we don't want to escape the metric in PJSIP, as that involves overhead that is wasted when either res_statsd isn't loaded or enabled. This patch takes an alternative approach. The Contact ID has been changed to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the aforementioned special characters, (b) can be done on Contact creation, which has minimal impact on run-time performance, and (c) also conforms to an earlier commit that changed the ID for dynamic contacts. The downside of this is that StatsD users will have to map SHA1 hashes back to the Contacts that are emitting the statistics. To that end, the CLI commands have been updated to include the first 10 characters of the MD5 hash, which should be enough to match what is shown in Graphite (or some other StatsD backend). ASTERISK-25595 #close Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2 Reported-by: Matt Jordan Tested-by: George Joseph 2015-11-30 22:19 +0000 [bd265a90be] gtjoseph * res_pjsip: Update logging to show contact->uri in messages An earlier commit changed the id of dynamic contacts to contain a hash instead of the uri. This patch updates status change logging to show the aor/uri instead of the id. This required adding the aor id to contact and contact_status and adding uri to contact_status. The aor id gets added to contact and contact_status in their allocators and the uri gets added to contact_status in pjsip_options when the contact_status is created or updated. ASTERISK-25598 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511 2015-12-01 16:11 +0000 [b5281b74e0] Jonathan Rose * Unset BRIDGEPEER when leaving a bridge Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da 2015-11-30 14:22 +0000 [59ba84e5cd] Richard Mudgett * res_sorcery_memory_cache.c: Fix off nominal ref leak. Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49 2015-11-30 16:42 +0000 [ef77439e39] Richard Mudgett * sched.c: Make not return a sched id of 0. According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 was never returned historically and several users incorrectly coded usage of the returned sched ID assuming that 0 was invalid. ASTERISK-25476 Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20 2015-11-25 12:23 +0000 [145d10a5d0] Richard Mudgett * Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions) chan_sip.c: * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to ao2 conversion. * Initialize register scheduler ids earlier because of ASTOBJ to ao2 conversion. chan_skinny.c: * Fix more scheduler usage for the valid 0 id value. ASTERISK-25476 Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95 2015-11-24 12:44 +0000 [fa20729032] Richard Mudgett * Audit improper usage of scheduler exposed by 5c713fdf18f. channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8 2015-12-01 07:55 +0000 [b24f2f4c2e] Alexander Traud * codec_resample: Increase buffer for Opus Codec. ASTERISK-25599 #close Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10 2015-11-30 11:13 +0000 [e5723d2776] gtjoseph * dns: Change lookup failures from LOG_ERROR to debug 1. dns.c and dns_system_resolver.c were spitting out errors for lookup failures for things like not finding a SRV record even though there was an A record. Those have been changed to debug messages. Logging not finding ANY record is left to the higher level caller. Also, dns_system_resolver was using Windows line endings so I converted them to Unix style. The actual log changes are on lines 156 and 159. Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094 2015-11-25 10:42 +0000 [270f7be54f] Alexander Traud * Build System: Support include-what-you-use. ASTERISK-25591 #close Change-Id: I8d3efa0826142ece9cbed2fd0d46f3b607fee6ae 2015-11-08 23:49 +0000 [f2a84b500d] Rodrigo Ramírez Norambuena * app_queue: Show reason of pause on CLI Add value of pause reason when is paused on CLI command "queue show" ASTERISK-25581 #close Report by: Rodrigo Ramírez Norambuena Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864 2015-11-27 07:39 +0000 [7cb8f2f33e] Niklas Larsson * CHANGES: Fix a typo Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7 2015-11-25 15:26 +0000 [9014f1f4a5] Kevin Harwell * fastagi: record file closed after sending result The fastagi record-file testsuite test sometimes fails reporting an empty recorded file. This was happening because Asterisk was sending the agi result notification prior to actually closing the file and the data, being buffered, had not been written to the file yet when the test attempts to check the file size. This patch makes it so the record file stream is closed prior to sending the agi result notification. ASTERISK-25593 #close Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde 2015-11-25 13:29 +0000 [03759c5587] Walter Doekes * main: Slight refactor of main. Improve color situation. Several issues are addressed here: - main() is large, and half of it is only used if we're not rasterisk; fixed by spliting up the daemon part into a separate function. - Call ast_term_init from rasterisk as well. - Remove duplicate code reading/writing asterisk history file. - Attempt to tackle background color issues and color changes that occur. Tested by starting asterisk -c until the colors stopped changing at odd locations. - Remove unused term_prep() and term_prompt() functions. ASTERISK-25585 #close Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f 2015-11-24 13:54 +0000 [91346b9fb7] David M. Lee * Fixed some typos Fixes some minor typos in the CHANGES file, plus an embarrasing typo in the StatsD API. Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7 2015-11-24 13:07 +0000 [fb45130476] Corey Farrell * res_pjsip_notify: Fix CLI usage info The usage info for 'pjsip send notify' previously referenced the chan_sip configuration sip_notify.conf. Fix this to reference the correct configuration pjsip_notify.conf. ASTERISK-25590 #close Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea 2015-11-18 09:43 +0000 [ee9c114747] Matt Jordan * res/res_endpoint_stats: Add module to emit endpoint StatsD statistics This patch adds a module that emits StatsD statistics about Asterisk endpoints. This includes: * A GAUGE statistic for endpoint states, tracking how many endpoints are in a particular state. * A GAUGE statistic for each endpoint, counting the number of channels currently associated with an endpoint. ASTERISK-25572 Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305 2015-11-23 14:27 +0000 [9ca652f1b9] Richard Mudgett * res_sorcery_realtime.c: Fix crash from NULL sorcery object type. If the sorcery object type is not found a NULL is returned. Unfortunately, sorcery_realtime_filter_objectset() will crash after complaining about not finding the object type and saying to expect errors. * Use ao2_cleanup() instead of ao2_ref() to prevent the crash. ASTERISK-25165 Reported by Corey Farrell Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97 2015-11-18 10:07 +0000 [75d90a9951] Matt Jordan * res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts This patch adds the ability to send StatsD statistics related to the state of PJSIP contacts. This includes: * A GUAGE statistic measuring the count of contacts in a particular state. This measures how many contacts are reachable, unreachable, etc. * The RTT time for each contact, if those contacts are qualified. This provides StatsD engines useful time-based data about each contact. ASTERISK-25571 Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c 2015-11-13 10:34 +0000 [482f2fc5ff] Matt Jordan * res/res_pjsip_outbound_registration: Add registration statistics for StatsD This patch adds outbound registration statistics for StatsD. This includes the following: * A GUAGE metric for the overall count of outbound registrations. * A GUAGE metric for each state an outbound registration can be in. As the outbound registrations change state, the overall count of how many outbound registrations are in the particular state is changed. These statistics are particularly useful for systems with a large number of SIP trunks, and where measuring the change in state of the trunks is useful for monitoring. ASTERISK-25571 Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37 2015-11-18 10:05 +0000 [97d7b344de] Matt Jordan * res_statsd: Add functions that support variable arguments Often, the metric names of statistics we are generating for StatsD have some dynamic component to them. This can be the name of a particular resource, or some internal status label in Asterisk. With the current set of functions, callers of the statsd API must first build the metric name themselves, then pass this to the API functions. This results in a large amount of boilerplate code and usage of either fixed length static buffers or dynamic memory allocation, neither of which is desireable. This patch adds two new functions to the StatsD API that support a printf style format specifier for constructing the metric name. A dynamic string, allocated in threadstorage, is used to build the metric name. This eases the burden on users of the StatsD API. Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea 2015-11-20 21:08 +0000 [726ee873a6] Matt Jordan * chan_pjsip: Handle T.38 faxes with direct media bridges When a channel is in a direct media bridge, a re-INVITE may arrive that forces Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge must change its technology to a simple bridge, and re-INVITE the media back to Asterisk. Generally, this logic mostly already exists in Asterisk. However, prior to this patch, there were a few bugs: (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from ever entering into a direct media bridge. This applies even when the only media being passed over the channel is audio. This patch fixes this bug by having the framehook specify that it defers caring about any frame type. This allows the channels to enter into a direct media bridge, which will be broken when a re-INVITE is received. (2) When a re-INVITE is received, nothing instructed the bridging layer to re-inspect the allowed bridging technology. This now occurs when either a re-INVITE is received from a peer, or when a response is received from the far end (that is, when the T.38 state changes to either T38_PEER_REINVITE or T38_LOCAL_REINVITE). (3) chan_pjsip needs to do a small amount of work to prevent a direct media bridge from being chosen when a T.38 session is in progress. When a T.38 session supplement has a t38 datastore - which is added when we detect we should start thinking about T.38 on a channel - we now refuse a native RTP bridge. (4) When a BYE request is received, we don't terminate the T.38 session. If the other side of a T.38 fax survives the hangup (due to the 'g' flag in Dial, for example), we don't currently re-INVITE the media on the other channel back to audio. This patch now has res_pjsip_t38 intercept BYE requests and inform the far side that the T.38 session is terminated. This naturally causes the correct re-INVITEs to be sent. ASTERISK-25582 Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb 2015-10-22 09:44 +0000 [9315a93757] Matt Jordan * main/cli: Use proper string methods to check existence of context/exten/app Because the context, extension, and application are stored in stringfields, checking for them being NULL doesn't work so well. This patch uses the appropriate string library call, ast_strlen_zero, to see if there is a value in the context/exten/app values. Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23 2015-11-20 21:07 +0000 [d2b141c79f] Matt Jordan * res/res_pjsip_t38: Add debug statements This patch adds some debug statements to res_pjsip_t38. These statements help to determine which SDP negotiation callbacks are being executed, and, when a particular callback exits, why a callback may not have applied its logic to the local or remote SDP. Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77 2015-11-19 09:40 +0000 [1bca90fcbe] Matt Jordan * res/res_pjsip_outbound_registration: Apply configuration on object type load When Asterisk is configured to use a dynamic sorcery backend (such as res_sorcery_astdb) with 'registration' objects, it will fail to create the internal state objects associated with the registration objects on module load. This is due to nothing actually querying for the specific objects and calling their sorcery apply handler during module load. This patch fixes that by calling get_registrations in the sorcery observer's object_type_loaded handler. Doing this causes the sorcery backends to be asked for the current state of all registration objects, which causes the apply handler to be called and the internal run-time state to be created. ASTERISK-25575 #close Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23 2015-11-11 06:29 +0000 [8ccb1d2bed] Alexander Traud * translate: Provide translation modules the result of SDP negotiation. Previously, a trancoding module did not have access to the joint but cached format. Therefore, the module did not have access to the attributes negotiated via SDP (line fmtp). Now, a translation module receives the joint format. ASTERISK-25545 #close Change-Id: Id6878a989b50573298dab115d3371ea369e1a718 2015-11-19 01:03 +0000 [92ea46ba94] Alexander Traud * res_format_attr_h264: Do not reset string buffer. When no parameter is present, Asterisk does not generate the line fmtp, as expected. However, because a buffer was reset, even rtpmap and fmtp of previous media codecs got removed. Now, Asterisk does not reset other codecs in case of no parameter for H.264. ASTERISK-25573 #close Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286 2015-11-18 02:25 +0000 [8c14b91651] Alec Davis * app_bridgeaddchan: ability to barge into existing call To be able to barge into a call by dialling a prefix+extension that maps to the extensions device. Senario is that DECT headset users may be away from their desks and need to transfer the call, the goal is that from any phone they dial a prefix then their extension and are added to the bridge that they are in, from there they can drop the headset call, as it's also on the handset, and transfer the caller. The dialplan would look like, where prefix=73, extension = 8512; exten => _738512,1,BridgeAdd(SIP/cisco0001) ASTERISK-25551 #close Reported By: Alec Davis Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540 2015-11-05 15:37 +0000 [05addf3d8f] Tyler Cambron * StatsD: Add sample rate compatibility Implemented support for the StatsD sample rate parameter, which is a parameter for determining when to send computed statistics to a client. Valid sample rate values are: Less than or equal to 0.0 will never be sent. Between 0.0 and 1.0 will randomly be sent. Greater than or equal to 1.0 will always be sent. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52 2015-11-17 14:53 +0000 [3dbaf696e9] Richard Mudgett * res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d 2015-11-17 14:53 +0000 [eaf898ac88] Richard Mudgett * res_pjsip_outbound_registration.c: Fix 423 response handling. Receiving a 423 Interval Too Brief response after authentication for an outbound registration attempt results in assuming that the registrar has rejected the registration permanently. If there are no configured retries for fatal responses then the outbound registration is stopped for that endpoint. For registrations, PJSIP/PJPROJECT intercepts the handling of 423 responses and does not include any authentication in the updated registration request. When the updated request is challenged then the Asterisk code assumes that we were challenged again because the peer rejected the authentication we sent earlier. * Made registration challenges keep track of the CSeq number to determine if the received challenge response was for the request we thought we sent. If the response's CSeq number differs from the CSeq number we last sent with authentication then authenticate again because it is a challenge to a different request. Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09 2015-11-18 00:20 +0000 [4013f9d577] Alec Davis * app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a 2015-11-16 16:10 +0000 [6919daab61] gtjoseph * dns: Fix pointer increment in dns_parse_answer_ex When dns_parse_answer_ex was iterating over the answers it wasn't incrementing the answer pointer correctly after the first answer. The result was that no answers after the first were being returned. For results where multiple records should have been sorted by priority, weight, etc., there was nothing to sort so the only the first record was returned even if it wouldn't have been the correct record based on the sort. ASTERISK-25565 #close Reported-by: Daniel Tryba Tested-by George Joseph Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b 2015-11-13 14:03 +0000 [ed13732188] Mark Michelson * Confbridge: Add a user timeout option This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98 2015-11-16 13:56 +0000 [a83e426e91] Matt Jordan * res/res_pjsip: Fix off nominal crash with requests that fail and have a timer When a request is sent using pjsip_endpt_send_request and fails, a condition exists where the request wrapper, which is an AO2 object, may be de-ref'd more times than it should. This occurs when the request's callback is called, and, in the callback, the timer on the PJSIP heap is cancelled. When that occurs, the request wrapper's lifetime is decremented. When pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of the request wrapper again, even though we've already cancelled the reference associated with the timer. This patch checks the return result of pj_timer_heap_cancel_if_active before removing the reference associated with the timer. We now only decrement it in this case if a timer is cancelled as a result of the function call. Change-Id: I21332343a1a019c1117076f9bf2df27be2850102 2015-11-14 07:02 +0000 [a1fcf6f7b2] Joshua Colp * hashtab: Add NULL check when destroying iterator. The hashtab API is pretty NULL tolerant which has resulted in remaining callers not doing much checks themselves. Unfortunately the function to destroy an iterator does not do a NULL check and will result in a crash if passed NULL. This change fixes that. ASTERISK-25552 #close Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619 2015-11-13 14:32 +0000 [436023a322] Richard Mudgett * res_pjsip_rfc3326.c: Fix crash when channel goes away. If an authenticated incoming caller does not respond to our 200 OK INVITE response with an ACK then PJSIP will hangup the call. Unfortunately, there is a chance that the session's channel will go away between one use of the channel pointer and another when building the BYE request because the BYE is being built by the monitor thread and not the call's serializer thread. * Added a check to ensure that the thread trying to add the Reason header is the call's serializer thread. This ensures that the channel will not go away on us. Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89 2015-11-13 14:19 +0000 [e8881e1770] Mark Michelson * Taskprocessors: Increase high-water mark In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce 2015-11-11 07:00 +0000 [fd23d423d8] Alexander Traud * format: Register format-attribute module with cached formats. In Asterisk 13, cached formats are created before their corresponding format- attribute module is registered. Cached formats are involved when a local extension is called. Therefore, ast_format_generate_sdp_fmtp did not work on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, and format-attribute modules provided externally. ASTERISK-25160 #close Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354 2015-11-12 11:17 +0000 [40b58a5d2b] Mark Michelson * res_pjsip distributor: Don't send 503 response to responses. When the SIP threadpool is backed up with tasks, we send 503 responses to ensure that we don't try to overload ourselves. The problem is that we were not insuring that we were not trying to send a 503 to an incoming SIP response. This change makes it so that we only send the 503 on incoming requests. Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404 2015-11-11 17:11 +0000 [264c74aa22] Mark Michelson * res_pjsip: Deny requests when threadpool queue is backed up. We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816 2015-11-12 06:24 +0000 [a159747660] Joshua Colp * format_cap: Don't append the 'none' format when appending all. When appending all formats of a type all the codecs are iterated and added. This operation was incorrectly adding the ast_format_none format which is special in that it is supposed to be used when no format is present. It shouldn't be appended. ASTERISK-25535 Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c 2015-11-11 04:16 +0000 [d982b99e71] Steve Davies * Further fixes to improper usage of scheduler When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b 2015-11-11 11:04 +0000 [2954354404] Joshua Colp * threadpool: Handle worker thread transitioning to dead when going active. This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143 2015-11-10 09:24 +0000 [525c7ab780] Alexander Traud * rtp_engine: Init a format-attribute module to its RFC defaults. Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52 2015-11-09 18:19 +0000 [be93036a4e] Corey Farrell * Remove ABI compatibility stub functions. ABI compatibility stubs existed for ast_app_separate_args and ast_verbose, this is not needed in master. Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453 2015-11-10 07:51 +0000 [02a124eda5] Corey Farrell * Remove execute permission from dns_system_resolver.c Change-Id: I3185735db42064bab00d3e073aed703385a00bf4 2015-11-09 03:01 +0000 [cf79b62778] Alexander Traud * ast_format_cap_get_names: To display all formats, the buffer was increased. ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a 2015-11-09 07:04 +0000 [e85f0c81af] Alexander Traud * ast_format_cap: Avoid format creation on module load, use cache instead. Since Asterisk 13, formats are immutable and cached. However while loading a module like chan_sip, some formats were created instead using cached ones. ASTERISK-25535 #close Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b 2015-11-06 07:54 +0000 [7dd8f89a50] Walter Doekes * func_callerid: Document that CALLERID(pres) is available. CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a 2015-11-06 07:52 +0000 [39daf9f066] Walter Doekes * docs: Fix a few typo's in app docs (more then, resourse). Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7 2015-11-06 14:19 +0000 [d82a4b098f] gtjoseph * dns: Use ntohl for ans->ttl in dns_parse_answer_ex dns_parse_answer_ex was not converting ans->ttl from network by order to host byte order which was causing certain ttls it to go negative. In turn this was causing answer edit checks to fail. ASTERISK-25528 #close Reported-by: Daniel Tryba Tested-by: George Joseph Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037 2015-11-06 07:36 +0000 [74e7333317] Walter Doekes * xmldoc: Improve xmldoc wrapping of 'core show ...' output. Previously, the wrapping did both lookahead and lookback, which, together with color escape sequences, caused some lines to be wrapped way earlier than other lines. This led to inconsistent output. This simplifies the wrapping code and makes it more sane: if maxcolumns is hit, we simply jump back to the last space and wrap there. ASTERISK-25527 #close Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957 2015-11-06 06:57 +0000 [9d6e917349] Sean Bright (license #5060) * res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP. In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual amount of channels is negotiated in-band. Therefore now, the Opus codec and its attribute rtpmap are registered with two channels. ASTERISK-24779 #close Reported by: PowerPBX Tested by: Alexander Traud patches: asterisk-24779.patch submitted by Sean Bright (license #5060) Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b 2015-11-03 16:19 +0000 [a2c2a8e1bb] Jonathan Rose * taskprocessor: Add high water mark warnings If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c 2015-11-02 20:11 +0000 [cd5ae02812] Corey Farrell * Increase account code maximum length to 80. This increases the maximum length of account code's to match extensions. This ensures it is always possible to set an accountcode to ${EXTEN} without truncation. ASTERISK-23904 Reported by: Ben Merrills Change-Id: If122602304ce03362722eb213a3111b32da5eeb9 2015-11-03 14:36 +0000 [379c041038] Tyler Cambron * StatsD: Add res_statsd compatibility Added a new api to res_statsd.c to allow it to receive a character pointer for the value argument. This allows for a '+' and a '-' to easily be sent with the value. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611 2015-11-04 14:31 +0000 [9c293b5104] Matt Jordan * main/dial: Protect access to the format_cap structure of the requesting channel When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f 2015-10-30 22:57 +0000 [b0bf189908] Corey Farrell * Fix cli display of build options. A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325 2015-11-03 10:58 +0000 [63e02b45c6] Matt Jordan * pjsip_configuration: On delete, remove the persistent version of an endpoint When an endpoint is deleted (such as through an API), the persistent endpoint currently continues to lurk around. While this isn't harmful from a memory consumption perspective - as all persistent endpoints are reclaimed on shutdown - it does cause Stasis endpoint related operations to continue to believe that the endpoint may or may not exist. This patch causes the persistent endpoint related to a PJSIP endpoint to be destroyed if the PJSIP endpoint is deleted. Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb 2015-11-03 11:15 +0000 [d33a1682e3] Matt Jordan * res_pjsip/location: Destroy contact_status objects on contact deletion The contact_status Sorcery objects are currently not destroyed when a contact is deleted. This causes the contact's last known RTT/status to be 'sticky' when the contact itself may no longer exist. This patch causes the contact_status objects associated with both dynamic and static contacts to be destroyed if the AoR holding those contacts is also destroyed (or via other paths where a contact may be deleted.) Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e 2015-11-03 08:15 +0000 [e26a06c1da] Matt Jordan * main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field The JSON packing for the ContactStatusChange event forgot to include the roundtrip_usec field. As a result, the field never showed up in any event, even when the data was available. This patch corrects that error by properly packing the JSON blob with the data. Change-Id: I8df80da659a44010afbd48f645967518ff5daa17 2015-11-02 20:24 +0000 [40574a2ea3] Corey Farrell * chan_sip: Allow websockets to be disabled. This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7 2015-11-02 17:19 +0000 [f80a0ae49b] Mark Michelson * res_pjsip: Set threadpool max size default to 50. During a stress test of subscriptions, a huge blast of subscription-related traffic resulted in the threadpool expanding to a ridiculous number of threads. The balooning of threads resulted in an increase of memory, which led to a crash due to being out of memory. An easy fix for the particular test was to limit the size of the threadpool, thus reining in the amount of memory that would be used. It was decided that there really is no downside to having a non-infinite default value for the maximum size of the threadpool, so this change introduces 50 threads as the maximum threadpool size for the SIP threadpool. ASTERISK-25513 #close Reported by John Bigelow Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be 2015-10-29 15:25 +0000 [c5093b21ad] Tyler Cambron * StatsD: Send stuff to the StatsD server and test Added code to allow the StatsD dialplan application to send data to the server specified in statsd.conf. ASTERISK-25419 Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922 2015-11-02 06:57 +0000 [014e3d426b] Matt Jordan * pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction When an AoR is created or destroyed dynamically, the scheduled OPTIONS requests that qualify the contacts on the AoR are not necessarily started or destroyed, particularly for persistent contacts created for that AoR. This patch adds create/update/delete sorcery observers for an AoR, which schedule/unschedule the qualifies as expected. Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d 2015-10-30 13:22 +0000 [80cf4960ff] Matt Jordan * Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs This patch adds a rule for installing the Super Awesome Company based 'Basic PBX' configuration files. As part of adding this rule, a bit of the content that makes up installing the configuration files under the 'samples' target was refactored into a make subroutine for usage by additional later config make targets. Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404 2015-10-29 08:28 +0000 [b522a5e30f] Joshua Colp * res_pjsip_pubsub: Fix assertion when UAS dialog creation fails. When compiled with assertions enabled one will occur when destroying the subscription tree when UAS dialog creation fails. This is because the code assumes that a dialog will always exist on a subscription tree when in reality during this specific scenario it won't. This change makes it so a dialog is not removed from the subscription tree if it is not present. ASTERISK-25505 #close Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee 2015-10-08 11:50 +0000 [fdfd0fb488] Tyler Cambron * StatsD: Add user input validation to the application Added code to accept user input and validate it before allowing it to be sent to the StatsD server. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3 2015-10-26 11:42 +0000 [d343a25173] Alexander Traud * chan_sip: Do not send all codecs on INVITE. Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 2015-10-19 07:11 +0000 [88f3dbaec9] Rodrigo Ramírez Norambuena * install_prereq: Update repositories before install on Debian systems When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa 2015-10-24 13:08 +0000 [4328d320c2] gtjoseph * build: GCC 5.1.x catches some new const, array bounds and missing paren issues Fixed 1 issue in each of the affected files. ASTERISK-25494 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77 2015-10-20 16:02 +0000 [a8aee0bbdb] gtjoseph * res_pjsip: Add "like" processing to pjsip list and show commands Add the ability to filter output from pjsip list and show commands using the "like" predicate like chan_sip. For endpoints, aors, auths, registrations, identifyies and transports, the modification was a simple change of an ast_sorcery_retrieve_by_fields call to ast_sorcery_retrieve_by_regex. For channels and contacts a little more work had to be done because neither of those objects are true sorcery objects. That was just removing the non-matching object from the final container. Of course, a little extra plumbing in the common pjsip_cli code was needed to parse the "like" and pass the regex to the get_container callbacks. Some of the get_container code in res_pjsip_endpoint_identifier was also refactored for simplicity. ASTERISK-25477 #close Reported by: Bryant Zimmerman Tested by: George Joseph Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1 2015-10-21 12:22 +0000 [691c0e0b31] Kevin Harwell * res_pjsip_outbound_registration: registration stops due to fatal 4xx response During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2 2015-10-22 17:07 +0000 [5dd9e1938a] Mark Michelson * format_cap: Detect vector allocation failures. A crash was seen on a system that ran out of memory due to Asterisk not checking for vector allocation failures in format_cap.c. With this change, if either of the AST_VECTOR_INIT calls fail, we will return a value indicating failure. Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 2015-10-02 15:32 +0000 [7f9823ff57] Mark Michelson * res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog. A certain situation can result in our attempting to send a NOTIFY on a destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but that subscriber has dropped off the network. We end up retransmitting that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY transaction. When the pjsip evsub code is told that the transaction has been terminated, it responds in kind by alerting us that the subscription has been terminated, destroying the subscription, and then removing its reference to the dialog, thus destroying the dialog. The problem is that when we get told that the subscription is being terminated, we detect that we have not sent a terminating NOTIFY request, so we queue up such a NOTIFY to be sent out. By the time that queued NOTIFY gets sent, the dialog has been destroyed, so attempting to send that NOTIFY can result in a crash. The fix being introduced here is actually a reintroduction of something the pubsub code used to employ. We hold a reference to the dialog and wait to decrement our reference to the dialog until our subscription tree object is destroyed. This way, we can send messages on the dialog even if the PJSIP evsub code wants to terminate earlier than we would like. In doing this, some NULL checks for subscription tree dialogs have been removed since NULL dialogs are no longer actually possible. Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5 2015-09-29 14:53 +0000 [e9e4bc9ece] Mark Michelson * res_pjsip_pubsub: Ensure dialog lock balance. When sending a NOTIFY, we lock the dialog and then unlock the dialog when finished. A recent change made it so that the subscription tree's dialog pointer will be set NULL when sending the final NOTIFY request out. This means that when we attempt to unlock the dialog, we pass a NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog remains locked after we think we have unlocked it. When a response to the NOTIFY arrives, the monitor thread attempts to lock the dialog, but it cannot because we never released the dialog lock. This results in Asterisk being unable to process incoming SIP traffic any longer. The fix in this patch is to use a local pointer to save off the pointer value of the subscription tree's dialog when locking and unlocking the dialog. This way, if the subscription tree's dialog pointer is NULLed out, the local pointer will still have point to the proper place and the dialog lock will be unlocked as we expect. Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a 2015-09-28 16:36 +0000 [b96267f7a3] Mark Michelson * res_pjsip_pubsub: Prevent crashes on final NOTIFY. The SIP dialog is removed from the subscription tree when the final NOTIFY is sent. However, after the final NOTIFY is sent, the persistence update function still attempts to access the cseq from the dialog, resulting in a crash. This fix removes the subscription persistence at the same time that the dialog is removed from the subscription tree. This way, there is no attempt to update persistence when the subscription is being destroyed. Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb 2015-09-17 17:28 +0000 [386cd7b2b0] Mark Michelson * res_pjsip_pubsub: Remove serializer when sending final NOTIFY. There have been crashes seen where a taskprocessor's listener is NULL unexpectedly. Looking at backtraces, the problem was specifically seen in PJSIP serializers. Subscriptions make the mistake of removing a serializer from a dialog during subscription tree destruction. Since subscription trees are reference-counted, guaranteeing the circumstances behind the destruction are not possible. This makes it so that the dialog serializer can be removed while not holding the dialog lock. This makes it possible for the distributor to get a pointer to the dialog serializer and have that serializer get freed out from under it. The fix for this is to remove the serializer from a subscription dialog when sending the final NOTIFY. This guarantees that the serializer is removed with the dialog lock held. By doing this, we guarantee that if the distributor gains access to the dialog's serializer, it will not be possible for the serializer to get freed by another thread. Change-Id: I21f5dac33529f65cec45679bdace60670800ff66 2015-09-02 09:14 +0000 [0b63d011c9] Mark Michelson * res_pjsip_pubsub: Fix crash on destruction of empty subscription tree. If an old persistent subscription is recreated but then immediately destroyed because it is out of date, the subscription tree will have no leaf subscriptions on it. This was resulting in a crash when attempting to destroy the subscription tree. A simple NULL check fixes this problem. Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac 2015-09-01 15:47 +0000 [ac0194dad6] Mark Michelson * res_pjsip_pubsub: Solidify lifetime and ownership of objects. There have been crashes and general instability seen in the pubsub code, so this patch introduces three changes to increase the stability. First, the ownership model for subscriptions has been modified. Due to RLS, subscriptions are stored in memory as a tree structure. Prior to my patch, the PJSIP subscription was the owner of the subscription tree. When the PJSIP subscription told us that it was terminating, we started destroying the subscription tree along with all of the individual leaf subscriptions that belong to the tree. The problem with this model is that the two actors in play here, the PJSIP subscription and the individual leaf subscriptions, need to have joint ownership of the subscription tree. So now, the PJSIP subscription and the individual leaf subscriptions each have a reference to the subscription tree. This way, we will not actually free memory until no players are left that care. The PJSIP subscription is a bigger stakeholder, in that if the PJSIP subscription's reference to the subscription tree is removed, the subscription tree instructs the leaf subscriptions to shut down and drop their references to the subscription tree when possible. The individual leaf subscriptions, upon being told to shut down, can drop their stasis subscriptions or whatever they use to learn of new state, and then drop their reference to the subscription tree once they are ready to die. Second, the lifetime of a PJSIP subscription's reference to our subscription tree has been altered. As I learned from doing a deep dive, the PJSIP evsub code can tell Asterisk multiple times that the subscription has been terminated, and not all of these times are especially helpful. I have altered the message flow that we use for SIP subscriptions such that we will always drop the PJSIP subscription's reference to the subscription tree when we send the NOTIFY that terminates a SIP subscription. This also means that we will now queue NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so that we can have predictable state changes from the PJSIP evsub code. Third, the synchronization of operations has been improved. PJSIP can call into our code from a serializer thread (e.g. upon receiving an incoming request) or from the monitor thread (e.g. when a subscription times out). Because of this, there is the possibility of competing threads stepping on each other. PJSIP attempts to do some synchronization on its own by always keeping the dialog lock held when it calls into us. However, since we end up pushing tasks into the serializer, the result was that serialized operations were not grabbing the dialog lock and could, as a result, step on something that was being attempted by a different thread. Now we ensure that serialized operations grab the dialog lock, then check for extenuating circumstances, then proceed with their operation if they can. Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5 2015-10-19 15:28 +0000 [1ce62b2545] Richard Mudgett * strings.c: Fix __ast_str_helper() to always return a terminated string. Users of functions which call __ast_str_helper() such as the ones listed below are likely to not check the return value for failure so ensuring that the string is always nil terminated is a good safety measure. ast_str_set_va() ast_str_append_va() ast_str_set() ast_str_append() Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 2015-10-19 15:27 +0000 [a04d946eaa] Richard Mudgett * Add missing failure checks to ast_str_set_va() callers. Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f 2015-10-21 11:44 +0000 [64c172deba] Joshua Colp * res_pjsip: Move URI validation to use time. In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a 2015-10-21 08:08 +0000 [f9cbac7321] Alexander Traud * format: Update the maximum packetization time for iLBC 30. In September 2006, the maximum packetization time (ptime) were set to such a low value, packetization was disabled for many codecs actually. This was fixed for many codecs but not for iLBC 30. This enables packetization for iLBC which can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. ASTERISK-7803 Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 2015-10-21 09:51 +0000 [f3b2b3d1b3] Alexander Traud * chan_sip: Fix autoframing=yes. With Asterisk 13, the structures ast_format and ast_codec changed. Because of that, the paketization timing (framing) of the RTP channel moved away from the formats/codecs. In the course of that change, the ptime of the callee was not honored anymore, when the optional autoframing was enabled. ASTERISK-25484 #close Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4 2015-10-20 22:24 +0000 [b425850f8b] Matt Jordan * rest-api-templates: Wikify error code response reasons Error response code descriptions may contain wiki markup that need to be escaped. Without this patch, Confluence will reject the document being sent and the responsible script will raise an exception. Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5 2015-10-20 12:06 +0000 [7be6194d6f] Matt Jordan * funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function When ab803ec342 was committed, it accidentally forgot to actually *add* the HOLD_INTERCEPT function. This highlights two interesting points: * Gerrit forces you to put the patch as it is going to into the repo up for review, which Review Board did not. Yay Gerrit. * No one apparently bothered to use this feature, or else they don't know about it. I'm going to go with the latter explanation. ASTERISK-24922 Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396 2015-10-19 14:14 +0000 [77780790e0] Jonh Wendell * main/cdr: Allow modules to modify CDR fields before dispatching them This patch adds the functions ast_cdr_modifier_register() ast_cdr_modifier_unregister() That work much like ast_cdr_register() and ast_cdr_unregister(). Modules registered will be given a chance to modify (or to do whatever they want) CDR fields just before they are passed to registered engines. Thus, for instance, if a module change the "userfield" field of a CDR, the modified value will be passed to every registered CDR backend for logging. ASTERISK-25479 #close Change-Id: If11d8fd19ef89b1a66ecacf1201e10fcf86ccd56 2015-10-19 19:59 +0000 [b9bd249a85] Matt Jordan * contrib/scripts/autosupport: Update for Asterisk 13 This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 (cherry picked from commit 9fc9777fa34753fb38991d42d8dbed516e907ca2) 2015-10-18 18:22 +0000 [92fa8d1e0e] Rodrigo Ramírez Norambuena * app_queue: Added reason pause of member In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e 2015-10-16 22:01 +0000 [b19860c03a] Corey Farrell * res_ari_events: Fix memory leak in mustache template. ASTERISK-25308 fixed a memory leak in res_ari_events.c, but this file is regenerated by a template and the template was not fixed. Change-Id: Ied4c6deae89d21f87f9cf99676b1d055aa83b38b 2015-10-14 14:15 +0000 [d799bcf361] mdu113 * res_config_pgsql.c: Fix deadlock loading realtime configuration. On v13, loading several thousand PJSIP endpoints on Asterisk start causes a deadlock most of the time. Thanks to mdu113 for discovering that there was a call to pgsql_exec() not protected by the pgsql_lock reentrancy lock. {quote} I believe a code path exists that attempts to use pgsql connection without locking pgsql_lock. I believe what happens during that deadlock that I see is two concurrent threads are both attempting to send query to pgsql, one of the thread is using a code path without locking pgsql_lock. If they managed to send queries at the same time, it seems postgres ignores one of the queries and replies only to the one of them. If it happens so that the thread holding the lock didn't receive the reply it will wait for it (and hold the lock) forever (or at least for very long time), thus completely blocking all access to db. {quote} * Added missing reentrancy locking around pgsql_exec() in find_table(). * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion between the psql_tables list lock and the pgsql_lock. ASTERISK-25455 #close Reported by: mdu113 Patches: res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113 Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2 2015-10-13 14:13 +0000 [13229037d1] Olle Johansson (License 5267) * channels/chan_sip: Set cause code to 44 on RTP timeout To quote Olle: "When issuing a hangup due to RTP timeouts the cause code is not set. I have selected 44 based on Cisco's implementation..." ASTERISK-25135 #close Reported by: Olle Johansson patches: rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc 2015-10-12 11:21 +0000 [984f100dab] Richard Mudgett * config.c: Fix off-nominal memory leak. Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 2015-10-12 11:20 +0000 [9951255775] Richard Mudgett * config.c: Fix potential memory corruption after [section](+). The memory corruption could happen if the [section](+) is the last section in the file with trailing comments. In this case process_text_line() has left *last_cat is set to newcat and newcat is destroyed. Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 2015-10-12 11:21 +0000 [c1ed11ee31] Richard Mudgett * config.c: Fix #include after [section](+). An #include right after a [section](+) would associate any variable assignments before a new section in the #include with the wrong section. * Fix section association by setting the current section to the appended section. * Fix '+' and '!' section flag interaction corner case depending upon which flag came first. If the '!' came first then it would be ignored. If the '!' came after then it would affect the appended section. The '!' will now no longer be ignored. ASTERISK-25461 #close Reported by: Sean Pimental Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 2015-10-10 15:20 +0000 [a12eb89ea4] Ivan Poddubny * Build: Add menuselect options for using compiler sanitizers This patch adds menuselect options for building Asterisk with various sanitizers provided by gcc and clang. When one of *SANITIZER flags is set in menuselect, the appropriate option is added to CFLAGS ad LDFLAGS for the build. Information on sanitizers in the project wiki: https://github.com/google/sanitizers/wiki GCC Manual: https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html Clang Compiler User's Manual: http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation ASTERISK-24718 #close Reported by: Badalian Vyacheslav Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0 2015-10-08 16:43 +0000 [ca030845ff] Richard Mudgett * configure: Fix check for libunbound to require v1.5.0 as minimum. Versions of libunbound before v1.4.21 do not compile with Asterisk. However, since v1.4.21 has a configure script bug that fails to detect the ldns library (which is fixed in v1.4.22) and v1.4.22 is not an easily detectable version we will require v1.5.0 as a minimum version of the library to work with Asterisk. ASTERISK-25108 #close Reported by: Richard Mudgett Change-Id: Ieb228bfb01467573fc121c7356a9dde27128894d 2015-10-08 11:50 +0000 [2fe9f09705] Tyler Cambron * StatsD: Write skeleton Asterisk application Wrote the skeleton framework for the Asterisk StatsD dialplan application. This includes a load function, unload function, a callback for execution, and XML documentation. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363 2015-10-06 18:01 +0000 [34d7fa6c4a] Richard Mudgett * res_pjsip: Fix deadlock when sending out-of-dialog requests. The struct send_request_wrapper has a pjsip lock associated with it that is created non-recursive. There is a code path for the struct send_request_wrapper lock that will attempt to lock it recursively. The reporter's deadlock showed that the thread calling endpt_send_request() deadlocked itself right after the wrapper object got created. Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited MWI NOTIFY messages can hit this deadlock. * Replaced the struct send_request_wrapper pjsip lock with the mutex lock that can come with an ao2 object since all of Asterisk's mutexes are recursive. Benefits include removal of code maintaining the pjsip non-recursive lock since ao2 objects already know how to maintain their own lock and the lock will show up in the CLI "core show locks" output. ASTERISK-25435 #close Reported by: Dmitriy Serov Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d 2015-10-06 11:05 +0000 [cc131832aa] Stefan Engström * res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending In ast_rtp_read, the value of the variable 'mark' which we try to assign to a frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate it to 0 or 1. ASTERISK-25451 #close Change-Id: I53bdf5c026041730184a6a809009c028549ce626 2015-10-07 01:24 +0000 [c944263e36] Ivan Poddubny * func_presencestate: Return "not_set" when no data is set in AstDB Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not exist, i.e. when a new CustomPresence is added in the dialplan. ASTERISK-25400 #close Reported by: Andrew Nagy Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a 2015-10-06 20:43 +0000 [4bf395e81e] Matt Jordan * res/res_rtp_asterisk: Fix assignment after ao2 decrement When we decide we will no longer schedule an RTCP write, we remove the reference to the RTP instance, then assign -1 to the stored scheduler ID in case something else comes along and wants to see if anything is scheduled. That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to fix the regression introduced by 3cf0f29310, this improper assignment on a potentially destroyed object started getting tripped on the build agents. Frankly, this should have been crashing a lot more often earlier. I can only assume that the timing was changed just enough by both changes to start actually hitting this problem. As it is, simply moving the assignment prior to the ao2 deference is sufficient to keep the RTP instance from being referenced when it is very, truly, aboslutely dead. (Note that it is still good practice to assign -1 to the scheduler ID when we know we won't be scheduling it again, as the ao2 deref *may* not always destroy the ao2 object.) ASTERISK-25449 Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7 2015-10-06 12:40 +0000 [3ec9cf7d6a] Florian Sauerteig * chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers. If a Via header containes an IPv6 address and a port number is ommitted, as it is the standard port, we now leave the port empty and to not set it to the value after the first colon of the IPv6 address. ASTERISK-25443 #close Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70 2015-10-05 16:53 +0000 [8fe9350b68] Richard Mudgett * chan_pjsip: Fix crash on reINVITE before initial INVITE completes. Apparently some endpoints attempt to send a reINVITE before completing the initial INVITE transaction. In this case PJSIP responds appropriately to the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip is using the initial INVITE transaction state to determine if an INVITE is the initial INVITE or a reINVITE. Since the initial INVITE transaction has not been confirmed yet chan_pjsip thinks the reINVITE is an initial INVITE and starts another PBX thread on the channel. The extra PBX thread ensures that hilarity ensues. * Fix checks for a reINVITE on incoming requests to look for the presence of a to-tag instead of the initial INVITE transaction state. * Made caller_id_incoming_request() determine what to do if there is a channel on the session or not. After a channel is created it is too late to just store the new party id on the session because the session's party id has already been copied to the channel's caller id. ASTERISK-25404 #close Reported by: Chet Stevens Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be 2015-10-05 21:34 +0000 [8cb614fe20] Matt Jordan * Fix improper usage of scheduler exposed by 5c713fdf18f When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of '0' returned. While this was valid per the documentation for the API, it was apparently never returned previously. As a result, several users of the scheduler API viewed the result as being invalid, causing them to reschedule already scheduled items or otherwise fail in interesting ways. This patch corrects the users such that they view '0' as valid, and a returned ID of -1 as being invalid. Note that the failing HEP RTCP tests now pass with this patch. These tests failed due to a duplicate scheduling of the RTCP transmissions. ASTERISK-25449 #close Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39 2015-08-26 16:58 +0000 [c6b0d60264] Debian Amtelco * chan_pjsip: Add Referred-By header to the PJSIP REFER packet. Some systems require the REFER packet to include a Referred-By header. If the channel variable SIPREFERREDBYHDR is set, it passes that value as the Referred-By header value. Otherwise, it adds the current dialog’s local info. Reported by: Dan Cropp Tested by: Dan Cropp Change-Id: I3d17912ce548667edf53cb549e88a25475eda245 2015-10-03 06:27 +0000 [89dec7675d] Ivan Poddubny * manager: Fix GetConfigJSON returning invalid JSON When GetConfigJSON was introduced back in 1.6, it returned each section as an array of strings: ["key=value", "key2=value2"]. Afterwards, it was changed a few times and became ["key": "value", "key2": "value2"], which is not a correct JSON. This patch fixes that by constructing a JSON object {} instead of an array []. Also, the keys "istemplate" and "tempates" that are used to indicate templates and their inherited categories are now wrapped in quotes. ASTERISK-25391 #close Reported by: Bojan Nemčić Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8 2015-09-30 17:28 +0000 [1b80dbeb60] Richard Mudgett * res_sorcery_memory_cache.c: Fix deadlock with scheduler. A deadlock can happen when a sorcery object is being expired from the memory cache when at the same time another object is being placed into the memory cache. There are a couple other variations on this theme that could cause the deadlock. Basically if an object is being expired from the sorcery memory cache at the same time as another thread tries to update the next object expiration timer the deadlock can happen. * Add a deadlock avoidance loop in expire_objects_from_cache() to check if someone is trying to remove the scheduler callback from the scheduler. ASTERISK-25441 #close Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc 2015-10-01 14:30 +0000 [9c1ca287a4] Richard Mudgett * res_sorcery_memory_cache.c: Replace inline code with function. Make sorcery_memory_cache_close() call remove_all_from_cache() instead of partially inlining it. ASTERISK-25441 Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c 2015-10-01 14:27 +0000 [6554a3b25e] Richard Mudgett * res_sorcery_memory_cache.c: Shutdown in a less crash potential order. Basically you should shutdown in the opposite order of how you setup since later setup pieces likely depend on earlier setup pieces. e.g., Registering your external API with the rest of the system should be the last thing setup and the first thing unregistered during shutdown. Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e 2015-09-30 17:27 +0000 [359394cc29] Richard Mudgett * res_sorcery_memory_cache.c: Misc tweaks. Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160 2015-09-30 17:27 +0000 [7942d1c2ff] Richard Mudgett * res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK. Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c 2015-09-30 13:42 +0000 [9f229d6a49] Joshua Colp * res_rtp_asterisk: Move "Set role" warning to be debug. In practice the set_role API callback can be invoked even when no ICE is present on an RTP instance. This can occur if ICE has not been enabled on it. ASTERISK-25438 #close Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69 2015-09-28 15:31 +0000 [9bc7386b7c] Richard Mudgett * sched.c: Add warning about negative time interval request. Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc 2015-09-25 18:37 +0000 [12feec0bf7] Richard Mudgett * res/ari/config.c: Fix user sort compare function. Made use the ao2 sort compare template function and OBJ_SEARCH_xxx identifiers. Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c 2015-09-25 17:26 +0000 [3f4fa245e5] Richard Mudgett * res/ari/config.c: Optimize conf_alloc() object init. * Now conf_alloc() has more off nominal error checking. * Eliminated RAII_VAR() use in conf_alloc(). * Eliminated a dubius shortcut when destroying cfg->general in conf_destructor() that would cause a crash if cfg->general failed to get allocated. * Add some ACO registration section comments. Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a 2015-09-25 16:48 +0000 [aa00df62ee] Richard Mudgett * res/ari/config.c: Fix conf_alloc() object init. Need to finish initializing the string fields in the ao2 object before putting any default strings into them. ASTERISK-25383 #close Reported by: yaron nahum Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84 2015-09-21 07:26 +0000 [2d7a4a3357] Matt Jordan * main/logger: Add log formatters and JSON structured logs When Asterisk is part of a larger distributed system, log files are often gathered using tools (such as logstash) that prefer to consume information and have it rendered using other tools (such as Kibana) that prefer a structured format, e.g., JSON. This patch adds support for JSON formatted logs by adding support for an optional log format specifier in Asterisk's logging subsystem. By adding a format specifier of '[json]': full => [json]debug,verbose,notice,warning,error Log messages will be output to the 'full' channel in the following format: { "hostname": Hostname or name specified in asterisk.conf "timestamp": Date/Time "identifiers": { "lwp": Thread ID, "callid": Call Identifier } "logmsg": { "location": { "filename": Name of the file that generated the log statement "function": Function that generated the log statement "line": Line number that called the logging function } "level": Log level, e.g., DEBUG, VERBOSE, etc. "message": Actual text of the log message } } ASTERISK-25425 #close Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238 2015-09-27 20:45 +0000 [9402f80726] Matt Jordan * res/res_stasis: Fix accidental subscription to 'all' bridge topic When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to all bridges. Unfortunately, the res_stasis control loop did not check that a bridge changing on a channel's control object was actually also non-NULL. As a result, app_subscribe_bridge will be called with a NULL bridge when a channel leaves a bridge. This causes a new subscription to be made to the bridge. If an application has also subscribed to the bridge, the application will now have two subscriptions: (1) The explicit one created by the app (2) The implicit one accidentally created by the control structure As a result, the 'BridgeDestroyed' event can be sent multiple times. This patch corrects the control loop such that it only subscribes an application to a new bridge if the bridge pointer is non-NULL. ASTERISK-24870 Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f 2015-09-04 13:51 +0000 [d6472d96b3] Scott Griepentrog * Scripts: check file versions of Asterisk and dependencies To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6 2015-09-24 14:56 +0000 [7c7a7ddd27] Richard Mudgett * app_queue.c: Force COLP update if outgoing channel name changed. * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 #close Reported by: John Hardin Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172 2015-09-24 14:20 +0000 [145608bd81] Richard Mudgett * app_queue.c: Factor out a connected line update routine. Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3 2015-09-24 13:27 +0000 [1d394774b2] Richard Mudgett * app_dial.c: Make 'A' option pass COLP updates. While the 'A' option is playing the announcement file allow the caller and peer to exchange COLP update frames. ASTERISK-25423 Reported by: John Hardin Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9 2015-09-24 12:59 +0000 [680b76eb25] Richard Mudgett * app_dial.c: Force COLP update if outgoing channel name changed. * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 Reported by: John Hardin Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c 2015-09-24 12:37 +0000 [fdf0bcb04a] Richard Mudgett * app_dial.c: Factor out a connected line update routine. Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091 2015-09-23 17:41 +0000 [c285879845] Richard Mudgett * app_dial.c: Remove some no-op code. Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54 2015-09-23 14:02 +0000 [3eefa07a39] Mark Michelson * logger: Prevent duplicate dynamic channels from being added. There was a problem observed where the "logger add channel" CLI command would allow for a channel with the same name to be added multiple times. This would result in each message being written out to the same file multiple times. The problem was due to the difference in how logger channel filenames are stored versus the format they are allowed to be presented when they are added. For instance, if adding the logger channel "foo" through the CLI, the result would be a logger channel with the file name /var/log/asterisk/foo being stored. So when trying to add another "foo" channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily add the duplicate channel. The fix presented here is to introduce two new methods in the logger code: * make_filename(): given a logger channel name, this creates the filename for that logger channel. * find_logchannel(): given a logger channel name, this calls make_filename() and then traverses the list of logchannels in order to find a match. This change has made use of make_filename() and find_logchannel() throughout to more consistently behave. ASTERISK-25305 #close Reported by Mark Michelson Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36 2015-09-24 14:49 +0000 [f42084be09] Mark Michelson * Do not swallow frames on channels leaving bridges. When leaving a bridge, indications on a channel could be swallowed by the internal indication logic because it appears that the channel is on its way to be hung up anyway. One such situation where this is detrimental is when channels on hold are redirected out of a bridge. The AST_CONTROL_UNHOLD indication from the bridging code is swallowed, leaving the channel in question to still appear to be on hold. The fix here is to modify the logic inside ast_indicate_data() to not drop the indication if the channel is simply leaving a bridge. This way, channels on hold redirected out of a bridge revert to their expected "in use" state after the redirection. ASTERISK-25418 #close Reported by Mark Michelson Change-Id: If6115204dfa0551c050974ee138fabd15f978949 2015-09-22 17:08 +0000 [06f4f80a63] Richard Mudgett * app_page.c: Fix crash when forwarding with a predial handler. Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981 2015-09-04 12:25 +0000 [b99a705262] Matt Jordan * ARI: Add the ability to subscribe to all events This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6 2015-09-21 18:06 +0000 [c74101509d] Kevin Harwell * app_record: RECORDED_FILE variable not being populated The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653 2015-09-21 08:16 +0000 [a29cf45