asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	9b7b8cb155	jansson: json_pack with new format to verify required runtime version. Add a json_pack at startup that will fail if runtime links against a library older than jansson-2.11. Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a	7 years ago
George Joseph	5f6890a8f9	Merge "res_stasis: Add ability to switch applications."	7 years ago
George Joseph	9390333862	Merge "Replace calls to strtok() with strtok_r()"	7 years ago
Friendly Automation	a502cb81db	Merge "samples: Fix comment typo in pjsip.conf.sample"	7 years ago
George Joseph	228205f520	Merge "bridging: Add creation timestamps"	7 years ago
Sean Bright	2473b791b9	Replace calls to strtok() with strtok_r() strtok() uses a static buffer, making it not thread safe. Also add a #define to cause a compile failure if strtok is used. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	7 years ago
Sean Bright	7b02a9617c	samples: Fix comment typo in pjsip.conf.sample Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857	7 years ago
Ben Ford	6626df586e	res_stasis: Add ability to switch applications. Added the ability to move between Stasis applications within Stasis. This can be done by calling 'move' in an application, providing (at minimum) the channel's id and the application to switch to. If the application is not registered or active, nothing will happen and the channel will remain in the current application, and an event will be triggered to let the application know that the move failed. The event name is "ApplicationMoveFailed", and provides the "destination" that the channel was attempting to move to, as well as the usual channel information. Optionally, a list of arguments can be passed to the function call for the receiving application. A full example of a 'move' call would look like this: client.channels.move(channelId, app, appArgs) The control object used to control the channel in Stasis can now switch which application it belongs to, rather than belonging to one Stasis application for its lifetime. This allows us to use the same control object instead of having to tear down the current one and create another. ASTERISK-28267 #close Change-Id: I43d12b10045a98a8d42541889b85695be26f288a	7 years ago
Friendly Automation	a32f525489	Merge "sip_to_pjsip: Make multiline comment parsing consistent with Asterisk"	7 years ago
Friendly Automation	5e21763560	Merge "app_queue: Handle empty 'interface' in queue member config"	7 years ago
Joshua Colp	c20b84e35b	Merge "res_pjsip_registrar: blocked threads on reliable transport shutdown take 3"	7 years ago
Joshua Colp	d7eed3bc5d	Merge "basic-pbx: Update configuration to work with current modules."	7 years ago
Sean Bright	f6b5b7208c	app_queue: Handle empty 'interface' in queue member config While the 'interface' column is a NOT NULL, the empty string is still allowed. res_config_odbc treats the empty string as a NULL and we crash when trying to dereference. Also cleaned up an adjacent error message for consistency. ASTERISK-28168 #close Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202	7 years ago
Sean Bright	f098d4a325	sip_to_pjsip: Make multiline comment parsing consistent with Asterisk In Asterisk configuration, a multiline comment starts with ;-- as long as it is not followed by another dash (i.e. ;--- is not a multiline comment). ASTERISK-28323 #close Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72	7 years ago
Friendly Automation	d490db8d58	Merge "res_pjsip_diversion: Use static pj_str_t for Diversion header names"	7 years ago
Joshua Colp	20e7ae1556	Merge "CHANGES: Document addition of 'wrapuptime' argument to AddQueueMember()"	7 years ago
Joshua Colp	2980622d2b	basic-pbx: Update configuration to work with current modules. The res_pjsip_websocket module requires the res_http_websocket module so ensure it is loaded. As well the res_pjsip_notify module needs the pjsip_notify.conf configuration file so ensure it is installed. ASTERISK-28272 Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd	7 years ago
sungtae kim	3638c433ac	bridging: Add creation timestamps This small feature will help to checking the bridge's status to figure out which bridge is in old/zombie or not. Also added detail items for the 'bridge show *' cli to provide more detail info. And added creation item to the ARI as well. ASTERISK-28279 Change-Id: I460238c488eca4d216b9176576211cb03286e040	7 years ago
Sean Bright	106a8ff05c	res_pjsip_diversion: Use static pj_str_t for Diversion header names PJSIP assumes that these header names are not allocated, and does not clone the name strings when reusing headers. Block unload of res_pjsip_diversion until shutdown to ensure static memory stays valid. ASTERISK-28312 #close Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9	7 years ago
Kevin Harwell	67d5c469fb	Merge "res_config_odbc: Avoid deadlock when max_connections = 1"	7 years ago
Rodrigo Ramírez Norambuena	f8295e0771	CHANGES: Document addition of 'wrapuptime' argument to AddQueueMember() Change-Id: Ieb332d018ae3f2fc82b9465381fde0f299af1611	7 years ago
Friendly Automation	3fbbd57338	Merge "menuselect: Add license header to menuselect_gtk.c"	7 years ago
Joshua Colp	a7e8a782a6	Merge "Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses.""	7 years ago
Sean Bright	8dc5f86095	menuselect: Add license header to menuselect_gtk.c This file was added to the Subversion repository on 2007-03-15 by Russell Bryant, a Digium employee at the time. ASTERISK-24173 #close Change-Id: Ie866fa9d31d550467613d362b35b03c031ee594d	7 years ago
Sean Bright	719a4643ab	res_config_odbc: Avoid deadlock when max_connections = 1 Rather than calling ast_odbc_find_table() in the prepare callback, call it beforehand and pass it in to the callback to avoid the need for a second connection. ASTERISK-28166 #close Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202	7 years ago
George Joseph	8f9ffe5905	res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream apply_negotiated_sdp_stream was returning a "1" when no joint capabilities were found on an outgoing call instead of a "-1". This indicated to res_pjsip_session that the handler DID handle the sdp when in fact it didn't. Without the appropriate setup, a subsequent media frame coming in would have an invalid stream_num and cause a seg fault when the stream was attempted to be retrieved. apply_negotiated_sdp_stream now returns the correct "-1" and any media is now discarded before it reaches the core stream processing. ASTERISK-28260 Reported by: Sotiris Ganouris Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f	7 years ago
Joshua Colp	47745fa1c9	Merge "res_pjsip_config_wizard: Don't crash if misconfigured"	7 years ago
Sean Bright	101272d0dc	Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses." This reverts commit `d524ad523d`. Reason for revert: This causes Contact and Via headers to have the wrong transport address. ASTERISK-28309 #close Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8	7 years ago
Friendly Automation	9d6161ee6a	Merge "res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen"	7 years ago
Sean Bright	82a43394ed	res_pjsip_config_wizard: Don't crash if misconfigured If both send_registrations and send_auth are both set to yes, outbound_auth/username must be set or we crash. ASTERISK-27992 #close Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d	7 years ago
Friendly Automation	b408bfdd06	Merge "res_mwi_devstate.c: New module to allow presence subs to VM boxes"	7 years ago
Kevin Harwell	930a7fe910	res_pjsip_registrar: blocked threads on reliable transport shutdown take 3 When a contact was removed by the registrar it did not always check to see if the circumstances involved a monitored reliable transport. For instance, if the 'remove_existing' option was set to 'true' then when existing contacts were removed due to 'max_contacts' being reached, those existing contacts being removed did not unregister the transport monitor. Also, it was possible to add more than one monitor on a reliable transport for a given aor and contact. This patch makes it so all contact removals done by the registrar also remove any associated transport monitors if necessary. It also makes it so duplicate monitors cannot be added for a given transport. ASTERISK-28213 Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a	7 years ago
George Joseph	e0fc663295	CI: Update jenkinsfiles with new Gerrit URLs The recent upgrade of Gerrit to 2.16 elimiated referencing a repository in a way the jenkinsfiles were relying on so the URL references were changed to a more consistent and supported format. Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e (cherry picked from commit `5ce084579f`)	7 years ago
Joshua C. Colp	f48a9c2520	Merge "rest-api-templates/asterisk_processor - replace http line breaks with line feed"	7 years ago
George Joseph	9ee76cf070	res_mwi_devstate.c: New module to allow presence subs to VM boxes This module allows presence subscriptions to voicemail boxes. This allows common BLF keys to act as voicemail waiting indicators. ASTERISK-28301 Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd	7 years ago
Torrey Searle	360f543677	res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen Delivery timeval in the smoother object will fall behind while a DTMF is being generated. This can eventually lead to invalid rtp timestamps. To prevent this from happening the smoother needs to be reset after every DTMF to keep the timing up to date. ASTERISK-28303 #close Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f	7 years ago
Friendly Automation	12c359ba5d	Merge "http.c: Support separated HTTP request"	7 years ago
Joshua C. Colp	18f92cb6f3	Merge "taskprocessor: Enable subsystems and overload by subsystem"	7 years ago
Friendly Automation	b37283e694	Merge "app_queue: Enable set the wrapuptime from AddQueueMember application"	7 years ago
Joshua C. Colp	5d08b4c463	Merge "res_ari_applications: Fix incorrect call to ao2_lock."	7 years ago
Joshua C. Colp	5f29d3b581	Merge "Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY"	7 years ago
Kevin Harwell	574128dec6	rest-api-templates/asterisk_processor - replace http line breaks with line feed Including line breaks (<br>, <br/>, <br />) in certain parts of the rest-api json definition (e.g. summary, notes) displays them correctly in swagger. However, when the field gets converted to the wiki format those breaks get escaped and show up in the text as the actual string literal "<br>" etc... This patch makes it so when converting to the wiki format it replaces all line break values (<br>, etc...) with line feeds ('\n'). Change-Id: Ie1c9faa0d1c5d622804cc0a21ce769095b08aa3d	7 years ago
Joshua C. Colp	e687cf214d	res_ari_applications: Fix incorrect call to ao2_lock. When listing the applications the apps lock was incorrectly locked twice instead of being locked and then unlocked. ASTERISK-28302 Change-Id: If7d064592a9e88c0f1049214c50e02be6dabf79e	7 years ago
Joshua Colp	e6b67b2a5d	res_pjsip_sdp_rtp: Allow only single ssrc attribute. When processing SSRC attributes we were iterating through all of them, even though we only need to know the remote SSRC once. This was problematic because some browsers group SSRCs together on a stream, and due to our negotiation only end up using the first one. Since we set the second one as the remote SSRC we would drop the received media from them instead of allowing it through. In the future this may be extended to allow SSRC groups and to use information from the attributes. Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270	7 years ago
Joshua C. Colp	88f77b3983	Merge "stasis: Store subscriber uniqueids with topic statistics."	7 years ago
Joshua C. Colp	cf1b64aaf0	Merge "res_pjsip_session Added rtcp stats result vector into the session"	7 years ago
Sungtae Kim	b4ccaad671	http.c: Support separated HTTP request Currently, the Asterisk does not support seperated HTTP request. This patch make the Asterisk enables to wait lest part of HTTP request. Also increases acceptable HTTP body length to 40k to support more larger request. ASTERISK-28236 Change-Id: I48a401aa64a21c3b37bf3cb4e0486d64b7dd8aa1	7 years ago
George Joseph	bc8dead610	Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY The current settings AST_PBX_MAX_STACK is 128 entries which is too low for some FreePBX installations with complex parking arrangements. Increased to 512 if LOW_MEMORY is not defined. ASTERISK-28300 Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299	7 years ago
Joshua C. Colp	a286f546f1	stasis: Store subscriber uniqueids with topic statistics. This change provides an easier mechanism to determine which subscribers are subscribed to a topic. Using this you can inspect the specific subscribers for further details. Change-Id: I8deea21703cd5c5357b85593b46c3eaf24e18c0c	7 years ago
George Joseph	c2adeb9dc2	taskprocessor: Enable subsystems and overload by subsystem To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	7 years ago

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