28288 Commits (e85adbd947fac6b5385f1b74eba506e53c78f483)

 

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Author	SHA1	Message	Date
Richard Mudgett	4286a369a1	res_pjsip: Whitespace and comment cleanup. ... Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	10 years ago
Corey Farrell	68de3a9e51	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	10 years ago
zuul	9473818659	Merge "res_srtp: Enable AES-256 and AES-GCM."	10 years ago
zuul	9372fe1b95	Merge "chan_dahdi.c: Fix deadlock potential in fax redirection."	10 years ago
zuul	a58f15ee4b	Merge "chan_sip.c: Fix deadlock potential in fax redirection."	10 years ago
zuul	ba2da66bd5	Merge "chan_pjsip.c: Fix deadlock potential in fax redirection."	10 years ago
zuul	3abf482393	Merge "res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook."	10 years ago
Joshua Colp	7f36b79f87	Merge "res_fax: Fix FAXOPT(faxdetect) timeout option."	10 years ago
Joshua Colp	4ffffa8bc4	Merge "chan_dahdi: Add faxdetect_timeout option."	10 years ago
Joshua Colp	0933f0cf96	Merge "res_pjsip: Add fax_detect_timeout endpoint option."	10 years ago
George Joseph	15bf6a87dc	Create Asterisk-14: Update CHANGES and UPGRADE files ... Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc	10 years ago
George Joseph	1b4922466b	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	10 years ago
zuul	194d0f606b	Merge "pbx: Create pbx_sw.c for management of 'struct ast_sw'."	10 years ago
zuul	fbdeb01edf	Merge "Add conditional support for noreturn functions."	10 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	10 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	10 years ago
Alexander Traud	1d2173c7ae	res_srtp: Enable AES-256 and AES-GCM. ... ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	10 years ago
zuul	3ca6407dab	Merge "Makefile: Retain XML Declaration and DTD in docs."	10 years ago
zuul	7ff9bed7b0	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only."	10 years ago
zuul	b1c45dc815	Merge "pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'."	10 years ago
zuul	e51b40bd87	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets."	10 years ago
zuul	b93c602198	Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts."	10 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	10 years ago
zuul	333a0fed33	Merge "Makefile: Suppress echoing of target 'config' again."	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 years ago
Joshua Colp	943bb48b59	Merge "pbx: Create pbx_include.c for management of 'struct ast_include'."	10 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	10 years ago
Alexander Traud	6428580e7f	Makefile: Suppress echoing of target 'config' again. ... ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f	10 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	10 years ago
zuul	26b4760808	Merge "app_queue: Only remove queue member from pending when state changes."	10 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	10 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	10 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	10 years ago
zuul	3cf33dd4e7	Merge "translate: explicit format destination not properly set"	10 years ago
zuul	afcc519dff	Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path."	10 years ago
zuul	707dbcbcd1	Merge "pbx: Fix leak of timezone for time based includes."	10 years ago
zuul	bea3e9b6fb	Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf."	10 years ago
Joshua Colp	89f0a7d3f4	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS."	10 years ago
zuul	f0f137e5bd	Merge "stasis_endpoint.c: Fix contactstatus_to_json()."	10 years ago
zuul	153875be24	Merge "pjsip_options.c: Fix container operation."	10 years ago
zuul	43596895f9	Merge "pjsip_configuration.c: Misc cleanups."	10 years ago