5390 Commits (7ec6a26dd6ea3e29740e2018229285eea8696afd)

Author	SHA1	Message	Date
Joshua Colp	7ec6a26dd6	Merge "sorcery.c: Tweak some container declaration formatting." into 14	9 years ago
Joshua Colp	251113408a	Merge "manager: Add <see-also> tags to relate AoC events and actions" into 14	9 years ago
Joshua Colp	8de930b593	Merge "manager: Add <see-also> links between related events" into 14	9 years ago
Joshua Colp	5d32617a25	Merge "pbx.c: Additional fixes to ast_context_remove_extension_callerid2." into 14	9 years ago
zuul	b8b9cee272	Merge "manager: Add <see-also> tags to relate UserEvent actions/apps/events" into 14	9 years ago
zuul	4b46e440df	Merge "manager: Add <see-also> tags to relate Bridge related events,actions, and apps" into 14	9 years ago
Richard Mudgett	aed60171f1	sorcery.c: Tweak some container declaration formatting. ... * Tweak sorcery_object_type_alloc() formatting. * Tweak ast_sorcery_init() formatting. Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944	9 years ago
Corey Farrell	f7f9081f7a	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	9 years ago
Matt Jordan	2e77b973fb	manager: Add <see-also> tags to relate interrelated events/actions together ... Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4	9 years ago
Matt Jordan	18065140d5	manager: Add <see-also> tags to relate Bridge related events,actions, and apps ... Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85	9 years ago
Matt Jordan	f69ae95889	manager: Add <see-also> tags to relate AoC events and actions ... Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e	9 years ago
Matt Jordan	c5574667c7	manager: Add <see-also> tags to relate UserEvent actions/apps/events ... Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4	9 years ago
Matt Jordan	3b3198333a	manager: Add <see-also> 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	9 years ago
zuul	906d0e72e1	Merge "Run mandatory cleanup when startup fails." into 14	9 years ago
zuul	d003057ff5	Merge "res_pjsip: Make aor named lock a mutex." into 14	9 years ago
Corey Farrell	03b80c5d27	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	9 years ago
zuul	6092514405	Merge "taskprocessor.c: Tweak high water checks." into 14	9 years ago
Richard Mudgett	7f9789fc39	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	9 years ago
Richard Mudgett	94ff684388	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	9 years ago
David M. Lee	919824ebbc	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	9 years ago
Joshua Colp	01c49ac889	Merge "res_srtp: Move SDP SRTP code from the core to res_srtp." into 14	9 years ago
Richard Mudgett	34d366eb0c	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	9 years ago
Alexei Gradinari	ecad35e147	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	9 years ago
Corey Farrell	bbaebd75c4	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	9 years ago
Corey Farrell	8e6a28e86b	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	9 years ago
Joshua Colp	44b1492c83	Merge "Remove SILK payload mappings from Asterisk core." into 14	9 years ago
zuul	62c3bb566c	Merge "pbx.c: Fix handling of '-' in extension name and callerid" into 14	9 years ago
Mark Michelson	d50895c7b0	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	9 years ago
zuul	a4771230fe	Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan." into 14	9 years ago
Joshua Colp	94a3c1f178	Merge "dsp.c: Add fax and DTMF detection unit tests." into 14	9 years ago
Joshua Colp	baa2629845	Merge "dsp.c: Added descriptive comments to Goertzel calculations." into 14	9 years ago
Joshua Colp	0dcbed074b	Merge "dsp.c: Fix incorrect format reference typo." into 14	9 years ago
Corey Farrell	5597193f62	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	9 years ago
Richard Mudgett	5316f49a2d	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	9 years ago
Kevin Harwell	e5529c4493	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	9 years ago
Richard Mudgett	5f54805e62	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	9 years ago
Richard Mudgett	ebc2b7695d	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	9 years ago
Richard Mudgett	a7e747918d	dsp.c: Fix incorrect format reference typo. ... Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	9 years ago
Richard Mudgett	ba449e0242	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	9 years ago
zuul	e7fc209b26	Merge "codecs: Add iLBC 20." into 14	9 years ago
George Joseph	58759bd77c	asterisk.c: Add auto generation and persistence of UUID ... Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d	9 years ago
Joshua Colp	804c3e6b9e	Merge "res_srtp: Enable AES-256 and AES-GCM." into 14	9 years ago
Alexander Traud	c82f24f36a	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	9 years ago
Richard Mudgett	d11731ac2f	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
zuul	8551830abc	Merge "pbx: Create pbx_sw.c for management of 'struct ast_sw'." into 14	9 years ago
Alexander Traud	2103ad1fec	res_srtp: Enable AES-256 and AES-GCM. ... ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	9 years ago
Corey Farrell	05cfe1a76e	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
Corey Farrell	0c88fb460f	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
zuul	b1c45dc815	Merge "pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'."	9 years ago
Joshua Colp	943bb48b59	Merge "pbx: Create pbx_include.c for management of 'struct ast_include'."	9 years ago