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Author	SHA1	Message	Date
Corey Farrell	8165115faa	res_smdi: Fix shutdown ref. ... When adding shutdown refs for OPTIONAL_API components I accidentally added it to the unload_module function in res_smdi. Move it to load_module. Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c	8 years ago
Sean Bright	8338f3ec14	res_hep: hepv3_is_loaded() should check if we are enabled ... res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of unnecessary work even if 'enabled' is set to 'no' in hep.conf. Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b	8 years ago
Jenkins2	7377927b18	Merge "pjsip: Ignore state changes from old transactions." into 13	8 years ago
Corey Farrell	85dec2ae4f	res_clialiases: Fix completion pass-through. ... Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8	8 years ago
Richard Mudgett	61e81338d9	res_rtp_asterisk.c: Disable packet flood detection for video streams. ... We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
Sean Bright	62e743e6a7	res_musiconhold: Start playlist after initial announcement ... Reset the samples counter to zero when we are done playing an announcement so that we don't skip into the middle of the first file in the playlist. Also add the selected annoucement to the output of 'moh show classes.' ASTERISK-24329 #close Reported by: Thomas Frederiksen Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0	8 years ago
Sean Bright	74073c395b	coverity: Fix warnings in res_smdi ... ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a	8 years ago
Sean Bright	a1fcb7b5a6	configs: Comment out and change IP of iax.conf [demo] ... This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100	8 years ago
Joshua Colp	9a016bd65e	Merge "menuselect: Tweak check for recently run configure." into 13	8 years ago
Joshua Colp	a35032816a	Merge "README: Remove outdated references to tex docs" into 13	8 years ago
Jenkins2	9a68fa36d1	Merge "res_pjsip: Assign support levels to a few modules" into 13	8 years ago
George Joseph	6d290f1880	README: Remove outdated references to tex docs ... Added links to the wiki to replace references to outdated tex docs. ASTERISK-27430 Reported by: Corey Farrell Change-Id: I5007e732b30bc7b63d124c530ae8857c89991209	8 years ago
Jenkins2	c35fa1868e	Merge "pjsip_options: contacts sometimes not being updated on reload" into 13	8 years ago
Jenkins2	53f1de6259	Merge "pjsip_options: dynamic contact's fields not updated on reload" into 13	8 years ago
Corey Farrell	dd72844e4d	CLI: Remove special handling of 'core set verbose' from rasterisk. ... rasterisk does not need to handle setting verbose levels locally, it should just tell the daemon what it wants and print what it is given. Just max out the verbose level on the local client so all filtering happens on the daemon. ASTERISK-20281 #close Change-Id: Ia305f75f1fc424a9169bfa30ef70d626ace2c8a8	8 years ago
Jenkins2	20ca06b1db	Merge "CLI: Fix 'core show sysinfo' function ordering." into 13	8 years ago
Jenkins2	992b7197b7	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)" into 13	8 years ago
George Joseph	e40bffd53b	Merge "AST-2017-012: Place single RTCP report block at beginning of report." into 13	8 years ago
Jenkins2	9b94e440ae	Merge "chan_sip: Don't crash in Dial on invalid destination" into 13	8 years ago
Joshua Colp	5705e8ae0e	AST-2017-012: Place single RTCP report block at beginning of report. ... When the RTCP code was transitioned over to Stasis a code change was made to keep track of how many reports are present. This count controlled where report blocks were placed in the RTCP report. If a compound RTCP packet was received this logic would incorrectly place a report block in the wrong location resulting in a write to an invalid location. This change removes this counting logic and always places the report block at the first position. If in the future multiple reports are supported the logic can be extended but for now keeping a count serves no purpose. ASTERISK-27382 ASTERISK-27429 Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116	8 years ago
Jenkins2	2b8a152c69	Merge "chan_sip: Don't send trailing \0 on keep alive packets" into 13	8 years ago
Joshua Colp	7a68faeb1d	pjsip: Ignore state changes from old transactions. ... When we fail over to a new target we create a new transaction and it becomes the current INVITE transaction. This does not prevent the previous transaction from raising state changes and causing the session to be prematurely disconnected if a transport error occurs immediately. This change backports a fix from PJSIP that eliminates the incorrect state change and reduces when they would be raised in the first place. ASTERISK-27408 Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34	8 years ago
Ivan Poddubny	7ffc41d19f	app_queue: Fix extension state subscriptions removed on dialplan reload ... The approach with having a single global subscription to all extension state changes has one issue: dynamically created hints don't have any watchers and are therefore garbage collected on the first dialplan reload. This change creates a state subscription for every queue member with a hint as state_interface, thus increasing the count of watches for hints, so they are not destroyed prematurely anymore. There are 2 side effects: 1. The state change callback in app_queue is not executed when there are no members referring to the extension. 2. The callback is called multiple times for the same hint if it's associated with more than one queue member. Reported by: Steven T. Wheeler ASTERISK-18411 #close Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89	8 years ago
Sean Bright	e1a358a6e4	chan_sip: Don't send trailing \0 on keep alive packets ... This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420	8 years ago
Sean Bright	ce2c89ce68	chan_sip: Don't crash in Dial on invalid destination ... Stripping the DNID in a SIP dial string can result in attempting to call the argument parsing macros on an empty string, causing a crash. ASTERISK-26131 #close Reported by: Dwayne Hubbard Patches: dw-asterisk-master-dnid-crash.patch (license #6257) patch uploaded by Dwayne Hubbard Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e	8 years ago
Corey Farrell	6632f61153	menuselect: Tweak check for recently run configure. ... Recently menuselect has randomly produced an error stating that configure was just run and make had to be restarted. I believe this is due to an incorrect menuselect/Makefile rule. The original rule produced an error if makeopts or autoconfig.h were older than makeopts.in or autoconfig.h.in. I believe this can create an issue if makeopts is older than autoconfig.h.in or if autoconfig.h is older than makeopts.in. The new rules compare files independently. Change-Id: Ibca155035fa1392c95e33cbf25f257902abba17b	8 years ago
Richard Mudgett	73b3390dbe	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) ... This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Sean Bright	ca448bf150	res_pjsip: Add TLSv1.1 and TLSv1.2 support ... Support for these protocols was added in the same commit as the 'proto' field, so we can safely use the same ./configure check. For reference: https://trac.pjsip.org/repos/changeset/4968 Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac	8 years ago
Sean Bright	d9b932a455	res_pjsip: Assign support levels to a few modules ... Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d	8 years ago
Corey Farrell	4decf4e492	CLI: Fix 'core show sysinfo' function ordering. ... Handle CLI initialization before any processing occurs. Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c	8 years ago
Richard Mudgett	28eddc7ea6	stasis_channels.c: Don't set channel snapshot caller_dnid twice. ... Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471	8 years ago
Jenkins2	60acbeb431	Merge "codec_opus: Make libcurl a dependency in menuselect" into 13	8 years ago
Jenkins2	3bfddd9e73	Merge "astdb: Improve prefix searches in astdb" into 13	8 years ago
Jenkins2	84fe5f0219	Merge "loader: Refactor resource_name_match." into 13	8 years ago
Joshua Colp	8834a8eff9	Merge "res_stasis and res_speech: Fix load order." into 13	8 years ago
Jenkins2	90fe599e45	Merge "utils: Add convenience function for setting fd flags" into 13	8 years ago
Jenkins2	3b6fc78f4e	Merge "pjsip: Improve CLI completion performance" into 13	8 years ago
Sean Bright	028f4320de	codec_opus: Make libcurl a dependency in menuselect ... ASTERISK-27475 #close Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800	8 years ago
Joshua Colp	ed089535d4	Merge "CDR: Fix deadlock setting some CDR values." into 13	8 years ago
Sean Bright	4838557132	pjsip: Improve CLI completion performance ... Use the new ast_cli_completion_add() function to improve completion performance for commands like 'pjsip show endpoint.' Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348	8 years ago
Sean Bright	a4b291029f	astdb: Improve prefix searches in astdb ... Using the LIKE operator requires a full table scan of 'astdb', whereas a comparison operation is able to use the primary key index. This patch adds a new function to the AstDB API for quick prefix matches and updates res_sorcery_astdb to utilize it. This showed substantial performance improvement in my test environment. Related to ASTERISK~26806, but does not completely resolve it. Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1	8 years ago
Corey Farrell	9fc4f10e0e	loader: Refactor resource_name_match. ... Optimize resource_name_match. This change eliminates use of ast_strdupa, instead verifying that both basename's are the same length, then using strncasecmp. Change-Id: I477275c0e954c99d74be5abfc8bb6545b04e5a3d	8 years ago
Sean Bright	c3bc44fa1b	pjsip_configuration: Add correct file header ... Change-Id: I25348c386a222bb704aff07f54375108a6402906	8 years ago
Sean Bright	f726f11974	utils: Add convenience function for setting fd flags ... There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Corey Farrell	3d79c34350	res_stasis and res_speech: Fix load order. ... res_stasis was missing AST_MODFLAG_LOAD_ORDER. Set res_stasis and res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for dependent modules. Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3	8 years ago
Kevin Harwell	ecdccb8071	pjsip_options: contacts sometimes not being updated on reload ... For both dynamic and static contacts it was possible that potential AOR changes were not being applied to all contacts. This was because the qualify and schedule code was only retrieving AOR's, and contacts with frequencies greater than zero. For instance the following could happen: and AOR/contact has a frequency of 5, it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are stopped, a list of AOR's is retrieved with frequency > 0, but none are selected since in this scenario all are 0. The contact for the one previously set to 5 though does not get updated, so it's status remains "AVAILABLE". This patch makes it so all contacts (static and dynamic) are selected, and appropriately updated if need be. ASTERISK-27467 #close Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb	8 years ago
Kevin Harwell	f20ab2b65f	pjsip_options: dynamic contact's fields not updated on reload ... Dynamic contacts were not being properly updated on reload. As a matter of fact any changes to the AOR that a dynamic contact was associated with were not being applied. On reload, this patch makes it so for each dynamic contact, the associated AOR is now retrieved and the AOR's fields are applied to the contact. ASTERISK-27467 Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d	8 years ago
Jenkins2	54a86779a3	Merge "translate: Skip matrix_rebuild during shutdown." into 13	8 years ago
Jenkins2	ac7501b15c	Merge "CLI: Fix remote console completion." into 13	8 years ago
Jenkins2	9bf73ae1b3	Merge "sounds_index: Avoid repeatedly reindexing." into 13	8 years ago