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Author	SHA1	Message	Date
zuul	ffbaefa48f	Merge "res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook." into 13	9 years ago
Joshua Colp	0b8448a74b	Merge changes from topic 'ASTERISK-26214' into 13 ... * changes: res_fax: Fix FAXOPT(faxdetect) timeout option. chan_dahdi: Add faxdetect_timeout option.	9 years ago
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." into 13	9 years ago
zuul	dec1e31f45	Merge "Add conditional support for noreturn functions." into 13	9 years ago
zuul	d4242d6250	Merge "Makefile: Retain XML Declaration and DTD in docs." into 13	9 years ago
zuul	7dacb14c03	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only." into 13	9 years ago
zuul	290269bb23	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets." into 13	9 years ago
zuul	7ce180a754	Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts." into 13	9 years ago
Corey Farrell	7fdf7c3d4c	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	9 years ago
Richard Mudgett	628e8c91d5	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	9 years ago
Richard Mudgett	676aeede36	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	9 years ago
Richard Mudgett	652130feb2	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	9 years ago
Richard Mudgett	851b1c3a17	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	9 years ago
Alexander Traud	021d4892cd	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	9 years ago
Corey Farrell	c8e41d14a1	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	9 years ago
Alexander Traud	e404f51b42	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	9 years ago
Alexander Traud	5f24874ebb	Makefile: Suppress echoing of target 'config' again. ... ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f	9 years ago
zuul	962c7ef5d9	Merge "app_queue: Only remove queue member from pending when state changes." into 13	9 years ago
Corey Farrell	76d4983c15	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	9 years ago
zuul	4b2031226d	Merge "Update support for SILK format." into 13	9 years ago
Alexei Gradinari	cb58f853e1	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	9 years ago
Mark Michelson	28501051b4	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	9 years ago
zuul	714ceb88df	Merge "translate: explicit format destination not properly set" into 13	9 years ago
Joshua Colp	8892029a03	Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path." into 13	9 years ago
zuul	b4976b9038	Merge "pbx: Fix leak of timezone for time based includes." into 13	9 years ago
zuul	64a09963be	Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf." into 13	9 years ago
zuul	b12aee68be	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS." into 13	9 years ago
zuul	3c1a40eb8d	Merge "stasis_endpoint.c: Fix contactstatus_to_json()." into 13	9 years ago
Joshua Colp	43b5f8d57b	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	9 years ago
zuul	56668e3e9c	Merge "pjsip_options.c: Fix container operation." into 13	9 years ago
zuul	91148fdd4f	Merge "pjsip_configuration.c: Misc cleanups." into 13	9 years ago
Corey Farrell	a17b071e36	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	9 years ago
zuul	436fc7ca2b	Merge "chan_sip: Fix reference leak in mwi_event_cb" into 13	9 years ago
zuul	8c3d301dc6	Merge "res/res_pjsip_session: Check for presence of an active negotiator" into 13	9 years ago
Joshua Colp	ca98b6cea2	Merge "res/res_pjsip_pubsub: Add additional debug statements" into 13	9 years ago
Joshua Colp	355fc081e6	Merge "res/res_corosync: Raise a Stasis message on node join/leave events" into 13	9 years ago
Kevin Harwell	8cef8f35e7	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	9 years ago
Richard Mudgett	afbd10b0c5	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	9 years ago
Corey Farrell	2be13d62fd	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	9 years ago
Alexander Traud	332beb27d8	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	9 years ago
Corey Farrell	672a64bda3	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	9 years ago
Richard Mudgett	fea201f7e6	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	9 years ago
Richard Mudgett	02877b4b4f	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	9 years ago
Alexander Traud	148cd1b319	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	9 years ago
zuul	8cea01ab1b	Merge "res_pjsip: Fix statsd regression." into 13	9 years ago
zuul	08f5f10f8f	Merge "BuildSystem: Allow own CFLAGS on ./configure." into 13	9 years ago
zuul	f5aa9f8034	Merge "install_prereq: Checkout of libSRTP 1.5.x." into 13	9 years ago
zuul	1cfe9514bd	Merge "chan_sip: Fix reference leaks in error paths." into 13	9 years ago
zuul	afd7aba585	Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed" into 13	9 years ago
zuul	daec52a8e6	Merge "res_pjsip: Added "subscribe_context" to endpoint" into 13	9 years ago