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Author	SHA1	Message	Date
Joshua C. Colp	783122ec87	Merge "res_musiconhold: Remove redundant option parsing"	7 years ago
Joshua C. Colp	b0db18432e	Merge "stasis: Allow empty application arguments to move."	7 years ago
Kevin Harwell	41dc6e57f6	Merge "app_queue: fix ring_entry to access nativeformats with a channel lock"	7 years ago
Kevin Harwell	8f5a6ed52d	Merge "chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function"	7 years ago
Joshua Colp	1d074debfb	stasis: Allow empty application arguments to move. ... Change-Id: I1e4d37415f3034abe36496dc30209c2303e6af5c	7 years ago
Joshua C. Colp	cb37486215	Merge "partial-inlining: disable partial-inlining if gcc>=8.2.1"	7 years ago
Dömsödi Gergely	48e407e506	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	7 years ago
George Joseph	6f158d27fc	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	7 years ago
Friendly Automation	4114c2bc3c	Merge "chan_pjsip: add a flag to ignore 183 responses if no SDP present"	7 years ago
Chris-Savinovich	449dff997c	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	7 years ago
George Joseph	30c19527bf	Merge "app_meetme: Don't mute joining admins if conference is muted"	7 years ago
Friendly Automation	d93c931503	Merge "res/res_rtp_asterisk.c: Fixing possible divide by zero"	7 years ago
cirillor	0d6d51b175	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	7 years ago
sungtae kim	8641fd9700	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	7 years ago
Sean Bright	825ea9ddb9	res_musiconhold: Remove redundant option parsing ... Change-Id: I481fabd8eaf2e4e7ffb5c8285b294742826e7d12	7 years ago
Torrey Searle	4661c08549	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	7 years ago
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	57850c7861	app_meetme: Don't mute joining admins if conference is muted ... ASTERISK-28328 #close Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b	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