asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	fbdeb01edf	Merge "Add conditional support for noreturn functions."	9 years ago
Corey Farrell	a36a174c4b	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	9 years ago
Alexei Gradinari	81ea024d93	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	9 years ago
Alexander Traud	1d2173c7ae	res_srtp: Enable AES-256 and AES-GCM. ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	9 years ago
zuul	3ca6407dab	Merge "Makefile: Retain XML Declaration and DTD in docs."	9 years ago
zuul	7ff9bed7b0	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only."	9 years ago
zuul	b1c45dc815	Merge "pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'."	9 years ago
zuul	e51b40bd87	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets."	9 years ago
zuul	b93c602198	Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts."	9 years ago
Corey Farrell	8f6e9ffcc6	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
zuul	333a0fed33	Merge "Makefile: Suppress echoing of target 'config' again."	9 years ago
Richard Mudgett	3d62f317dd	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	9 years ago
Richard Mudgett	db4979fa79	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	9 years ago
Richard Mudgett	3db468ea9e	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	9 years ago
Richard Mudgett	9abbea162c	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	804fbd9c2b	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	0d1744e132	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	e739888d99	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	6fca2b3bf0	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	cf1188a1be	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
Alexei Gradinari	e9daa34261	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
Joshua Colp	943bb48b59	Merge "pbx: Create pbx_include.c for management of 'struct ast_include'."	9 years ago
Alexander Traud	cb5e3445be	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	6428580e7f	Makefile: Suppress echoing of target 'config' again. ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f	9 years ago
Corey Farrell	e2e8713b84	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	9 years ago
Alexander Traud	d56fc3b36b	translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ASTERISK-25629 #close Change-Id: I66c0086e6c17764b8141ec60a3e2aaefe088eb78	9 years ago
zuul	26b4760808	Merge "app_queue: Only remove queue member from pending when state changes."	9 years ago
Corey Farrell	be36bd7ca5	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	9 years ago
Corey Farrell	d3348c51b5	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
Mark Michelson	273052f404	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	3cf33dd4e7	Merge "translate: explicit format destination not properly set"	9 years ago
zuul	afcc519dff	Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path."	9 years ago
zuul	707dbcbcd1	Merge "pbx: Fix leak of timezone for time based includes."	9 years ago
zuul	bea3e9b6fb	Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf."	9 years ago
Joshua Colp	89f0a7d3f4	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS."	9 years ago
zuul	f0f137e5bd	Merge "stasis_endpoint.c: Fix contactstatus_to_json()."	9 years ago
zuul	153875be24	Merge "pjsip_options.c: Fix container operation."	9 years ago
zuul	43596895f9	Merge "pjsip_configuration.c: Misc cleanups."	9 years ago
Joshua Colp	31967dacdf	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
Joshua Colp	5703374cc5	Merge "chan_sip: Fix reference leak in mwi_event_cb"	9 years ago
Corey Farrell	f3608b50d7	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	2567e57624	Merge "res/res_corosync: Raise a Stasis message on node join/leave events"	9 years ago
zuul	3849f23bff	Merge "res/res_pjsip_session: Check for presence of an active negotiator"	9 years ago
Kevin Harwell	63ac4c9487	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	2f26512fd8	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
Richard Mudgett	bc1ff41be7	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	eabcfeeaa3	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
Corey Farrell	f73ddde7d4	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
Corey Farrell	fd54d69feb	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
Alexander Traud	85212f2799	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

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