asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	455bc78901	Merge "Refactor usage pattern of xmldoc info tag." into 13	9 years ago
Joshua Colp	d85310e3e8	Merge "sorcery.c: Minor optimizations." into 13	9 years ago
zuul	7ed9bc8169	Merge "sorcery.c: Tweak some container declaration formatting." into 13	9 years ago
zuul	050b90769a	Merge "core: Entity ID is not set or invalid" into 13	9 years ago
zuul	6c4c31456f	Merge "pbx.c: Additional fixes to ast_context_remove_extension_callerid2." into 13	9 years ago
Corey Farrell	f4e28b3a09	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	9 years ago
Joshua Colp	10fa55576d	Merge "manager: Add <see-also> tags to relate interrelated events/actions together" into 13	9 years ago
Joshua Colp	952edd30d1	Merge "manager: Add <see-also> tags to relate Bridge related events,actions, and apps" into 13	9 years ago
Joshua Colp	b4828cb125	Merge "manager: Add <see-also> tags to relate AoC events and actions" into 13	9 years ago
Joshua Colp	274794eb1e	Merge "manager: Add <see-also> tags to relate UserEvent actions/apps/events" into 13	9 years ago
zuul	ebdc880eb3	Merge "manager: Add <see-also> links between related events" into 13	9 years ago
Richard Mudgett	74a91b9ee5	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	9 years ago
Richard Mudgett	29beb2890c	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	9b822293bd	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
Alexei Gradinari	403c794684	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	9 years ago
Matt Jordan	4facaac408	manager: Add <see-also> tags to relate interrelated events/actions together Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4	9 years ago
Matt Jordan	232d4fe24f	manager: Add <see-also> tags to relate Bridge related events,actions, and apps Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85	9 years ago
Matt Jordan	63c0b2f7c9	manager: Add <see-also> tags to relate AoC events and actions Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e	9 years ago
Matt Jordan	0422667d6c	manager: Add <see-also> tags to relate UserEvent actions/apps/events Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4	9 years ago
Matt Jordan	781bb410d0	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	03b7d04298	Merge "Run mandatory cleanup when startup fails." into 13	9 years ago
zuul	4a05856f40	Merge "res_pjsip: Make aor named lock a mutex." into 13	9 years ago
Corey Farrell	cb043249b6	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
Richard Mudgett	9d4bd3d763	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	e1248c3075	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
Alexei Gradinari	1589452fdc	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	805f105f88	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
zuul	9db6cdba1a	Merge "asterisk.c: Add auto generation and persistence of UUID" into 13	9 years ago
George Joseph	ad3e65433c	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
Mark Michelson	f6821fbaec	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 (cherry picked from commit `d50895c7b0`)	9 years ago
zuul	948a9b615f	Merge "pbx.c: Fix handling of '-' in extension name and callerid" into 13	9 years ago
zuul	c2aed1c6e0	Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan." into 13	9 years ago
Joshua Colp	cc5660406c	Merge "dsp.c: Add fax and DTMF detection unit tests." into 13	9 years ago
Joshua Colp	cc825e54b0	Merge "dsp.c: Added descriptive comments to Goertzel calculations." into 13	9 years ago
Joshua Colp	323042d99d	Merge "dsp.c: Fix incorrect format reference typo." into 13	9 years ago
Corey Farrell	57e9c66819	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	873fc0fda5	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	852e763571	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	e8c34680ca	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	c1f240b818	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	003a52fd62	dsp.c: Fix incorrect format reference typo. Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	9 years ago
Richard Mudgett	159e437e5a	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
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." 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	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
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
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

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5256 Commits (2381ddde63489b8601c9514111838fb4656610b4)