asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	03c0d19cc8	stasis: Allow empty application arguments to move. Change-Id: I1e4d37415f3034abe36496dc30209c2303e6af5c	6 years ago
Joshua C. Colp	c8a1f8bfaa	Merge "partial-inlining: disable partial-inlining if gcc>=8.2.1" into 16	6 years ago
Kevin Harwell	6a435723f7	AMI/ARI: Bump non-breaking version numbers Increasing the non-breaking AMI and ARI version numbers due to changes and additions in those API's. Note, some changes may be forthcoming (will be added between now and the next release of Asterisk), thus not listed here. As well a few changes listed below may have been released in a previous release of Asterisk, but the API version numbers were not increased at that time, so including here. AMI: * app_queue: set the wrapuptime from AddQueueMember application - `e806990` * res_pjsip: option for ContactStatus event updates - `7f22c9f` ARI: * bridging: Add creation timestamps - `0d70120` * res_stasis: Add ability to switch applications - `50a4b61` * ARI event type filtering - `1c5def4` * Added ARI resource /ari/asterisk/ping - `19fc99a` ASTERISK-28314 Change-Id: Iebc813840f8230afa6b20579772e15549064b787	6 years ago
George Joseph	a71ae3dc44	Makefile.moddir_rules: Pass PJPROJECT_BUNDLED to download_externals The download_externals script wasn't getting the PJPROJECT_BUNDLED environment variable passed down to it so it wasn't downloading the appropriate variant of res_digium_phone. This could cause crashes in the DPMA. Change-Id: I5daa9369c7af1fd556d892e89a85f279a2533425	6 years ago
Chris-Savinovich	47c853f8e1	partial-inlining: disable partial-inlining if gcc>=8.2.1 Apply flag -fno-partial-inlining on default optimization if and only if gcc version >= 8.2.1 (this is the current ver on Fedora and Ubuntu). This is done to avoid a bug that causes arithmetic calculations to fail if the following conditions are met: 1. TEST_FRAMEWORK on 2. DONT_OPTIMIZE off 3. Fedora and Ubuntu 4. GCC 8.2.1 5. There must exist a certain combination of multithreading. 6. Optimization level -O2 and -O3 7. Flag -fpartial-inline activated (default when optimization level>=2) The following link points to a similar gcc bug reported in 2015. This leads me to believe the bug has regressed. Note I am not able to replicate this bug in an environment other than Asterisk + Test Framework + Test_cel because the multithreading combination that causes it seems to be unique. Therefore I am temporarily abandoning any thoughts of reporting the new occurrence of this bug to gcc.gnu.org. https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65307 Change-Id: Ibd1afe60e0a38b88e85fdcd9b051004601c2f102	6 years ago
Friendly Automation	b428363cd5	Merge "res/res_rtp_asterisk.c: Fixing possible divide by zero" into 16	6 years ago
George Joseph	9b15e04011	Merge "jansson: json_pack with new format to verify required runtime version." into 16	6 years ago
Joshua Colp	b886fef66e	stasis: Improve topic/subscription names and statistics. Topic names now follow: <subsystem>:<functionality>[/<object>] This ensures that they are all unique, and also provides better insight in to what each topic is for. Subscriber ids now also use the main topic name they are subscribed to and an incrementing integer as their identifier to make it easier to understand what the subscription is primarily responsible for. Both the CLI commands for listing topic and subscription statistics now sort to make it a bit easier to see what is going on. Subscriptions will now show all topics that they are receiving messages from, not just the main topic they were subscribed to. ASTERISK-28335 Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d	6 years ago
Friendly Automation	13cca7bb11	Merge "app_meetme: Don't mute joining admins if conference is muted" into 16	6 years ago
sungtae kim	ca8f254f0f	res/res_rtp_asterisk.c: Fixing possible divide by zero Currently, when the Asterisk calculates rtp statistics, it uses sample_count as a unsigned integer parameter. This would be fine for most of cases, but in case of large enough number of sample_count, this might be causing the divide by zero error. ASTERISK-28321 Change-Id: If7e0629abaceddd2166eb012456c53033ea26249	6 years ago
cirillor	c9459b1875	chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function. ASTERISK-28317 Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a	6 years ago
Sean Bright	8ca74d73f1	res_musiconhold: Remove redundant option parsing Change-Id: I481fabd8eaf2e4e7ffb5c8285b294742826e7d12	6 years ago
Torrey Searle	4ca41caf38	chan_pjsip: add a flag to ignore 183 responses if no SDP present chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	6 years ago
Corey Farrell	641b6041c7	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	6 years ago
George Joseph	255e374254	Merge "samples: Fix comment typo in pjsip.conf.sample" into 16	6 years ago
Friendly Automation	70cd169fb1	Merge "Replace calls to strtok() with strtok_r()" into 16	6 years ago
Friendly Automation	afe16021ef	Merge "res_stasis: Add ability to switch applications." into 16	6 years ago
Friendly Automation	7566d9bb61	Merge "bridging: Add creation timestamps" into 16	6 years ago
Sean Bright	5b5aa05bd9	app_meetme: Don't mute joining admins if conference is muted ASTERISK-28328 #close Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b	6 years ago
Sean Bright	4860954540	Replace calls to strtok() with strtok_r() strtok() uses a static buffer, making it not thread safe. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	6 years ago
Sean Bright	90f06ad169	samples: Fix comment typo in pjsip.conf.sample Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857	6 years ago
Ben Ford	50a4b618fc	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	6 years ago
Dömsödi Gergely	e7dd4bd4a9	app_queue: fix ring_entry to access nativeformats with a channel lock Fixes an intermittent segmentation fault which occured when accessing nativeformats of a channel which entered into a queue. ASTERISK-27964 Reported by: Francisco Seratti Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e	6 years ago
Joshua Colp	24b94aa27f	Merge "sip_to_pjsip: Make multiline comment parsing consistent with Asterisk" into 16	6 years ago
Joshua Colp	293df3c7ac	Merge "app_queue: Handle empty 'interface' in queue member config" into 16	6 years ago
Joshua Colp	f9bc474ab0	Merge "res_pjsip_registrar: blocked threads on reliable transport shutdown take 3" into 16	6 years ago
Joshua Colp	7bdbccc64a	Merge "basic-pbx: Update configuration to work with current modules." into 16	6 years ago
Sean Bright	88681ba9bb	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	6 years ago
Sean Bright	0cfafb89b5	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	6 years ago
Joshua Colp	90f0f6659d	Merge "res_pjsip_diversion: Use static pj_str_t for Diversion header names" into 16	6 years ago
Joshua Colp	6a714c0e9f	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	6 years ago
sungtae kim	0d7012044a	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	6 years ago
Rodrigo Ramírez Norambuena	63a3321c46	CHANGES: Document addition of 'wrapuptime' argument to AddQueueMember() Change-Id: Ieb332d018ae3f2fc82b9465381fde0f299af1611	6 years ago
Sean Bright	ffe07772e6	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	6 years ago
Friendly Automation	2a387f04db	Merge "res_config_odbc: Avoid deadlock when max_connections = 1" into 16	6 years ago
Kevin Harwell	436975dd6e	Merge "menuselect: Add license header to menuselect_gtk.c" into 16	6 years ago
Joshua Colp	a08ee6c962	Merge "Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses."" into 16	6 years ago
Sean Bright	625af27a28	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	6 years ago
George Joseph	61585af9bd	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	6 years ago
Joshua Colp	1d7df1d3f1	Merge "res_pjsip_config_wizard: Don't crash if misconfigured" into 16	6 years ago
Sean Bright	6f1c0e787c	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	6 years ago
Joshua Colp	ae1b12b834	Merge "res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen" into 16	6 years ago
Joshua Colp	c268b3592c	Merge "res_mwi_devstate.c: New module to allow presence subs to VM boxes" into 16	6 years ago
Sean Bright	fe694de299	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	6 years ago
Sean Bright	edf31624a2	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	6 years ago
Kevin Harwell	b3ebb51086	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	6 years ago
George Joseph	6e8cba39a4	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`)	6 years ago
Joshua C. Colp	cc8aa4eee1	Merge "rest-api-templates/asterisk_processor - replace http line breaks with line feed" into 16	6 years ago
George Joseph	8402189892	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	6 years ago
Torrey Searle	5b8db4e422	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	6 years ago

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