asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	8a281776d5	Merge "pjsip_options: dynamic contact's fields not updated on reload"	8 years ago
George Joseph	afb6e85980	Merge "CLI: Fix 'core show sysinfo' function ordering."	8 years ago
Joshua Colp	c50905756b	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"	8 years ago
Joshua Colp	62f2860c39	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	58514c7442	Merge "chan_sip: Don't crash in Dial on invalid destination"	8 years ago
Jenkins2	e7dccbe708	Merge "chan_sip: Don't send trailing \0 on keep alive packets"	8 years ago
Sean Bright	0c9cc7e975	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	5039b5741c	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
Richard Mudgett	22810fc635	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	ec1f4bf48d	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
Corey Farrell	c01ba7437e	CLI: Fix 'core show sysinfo' function ordering. Handle CLI initialization before any processing occurs. Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c	8 years ago
Richard Mudgett	c2ec82bf36	stasis_channels.c: Don't set channel snapshot caller_dnid twice. Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471	8 years ago
Jenkins2	38dcdf2d68	Merge "codec_opus: Make libcurl a dependency in menuselect"	8 years ago
Jenkins2	b7e79d7baf	Merge "astdb: Improve prefix searches in astdb"	8 years ago
Jenkins2	f81a884033	Merge "loader: Refactor resource_name_match."	8 years ago
Joshua Colp	5d43a4d4ff	Merge "res_stasis and res_speech: Fix load order."	8 years ago
Jenkins2	710b3a29c3	Merge "utils: Add convenience function for setting fd flags"	8 years ago
Sean Bright	00578fae0a	codec_opus: Make libcurl a dependency in menuselect ASTERISK-27475 #close Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800	8 years ago
Jenkins2	cd741136c5	Merge "pjsip: Improve CLI completion performance"	8 years ago
Jenkins2	6d25b26933	Merge "CDR: Fix deadlock setting some CDR values."	8 years ago
Sean Bright	521f741b04	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	9a9edc6c9e	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	d2e87b8e14	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	dbb376f166	pjsip_configuration: Add correct file header Change-Id: I25348c386a222bb704aff07f54375108a6402906	8 years ago
Sean Bright	2ffe52a116	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	e2dbc26376	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	bd2218ce63	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	b0b28446c1	Merge "translate: Skip matrix_rebuild during shutdown."	8 years ago
George Joseph	d7f3f8c283	Merge "sounds_index: Avoid repeatedly reindexing."	8 years ago
George Joseph	5cef8b2f3e	Merge "media_index: Improve startup."	8 years ago
Corey Farrell	c2c9995830	translate: Skip matrix_rebuild during shutdown. Change-Id: I1e5eef4029cba56e33d786c5a5ade8091e531a1e	8 years ago
Corey Farrell	ab191e9782	sounds_index: Avoid repeatedly reindexing. The sounds index is rebuilt each time a format is registered or unregistered. This causes the index to be repeatedly rebuilt during startup and shutdown. This patch significantly reduces the work done by delaying sound index initialization until after modules are loaded. This way a reindex only occurs if a format module is loaded after startup. We also skip reindexing when format modules are unloaded during shutdown. Change-Id: I585fd6ee04200612ab1490dc804f76805f89cf0a	8 years ago
Richard Mudgett	3078b7adc2	CDR: Fix deadlock setting some CDR values. Setting channel variables with the AMI Originate action caused a deadlock when you set CDR(amaflags) or CDR(accountcode). This path has the channel locked when the CDR function is called. The CDR function then synchronously passes the job to a stasis thread. The stasis handling function then attempts to lock the channel. Deadlock results. * Avoid deadlock by making the CDR function handle setting amaflags and accountcode directly on the channel rather than passing it off to the CDR processing code under a stasis thread to do it. * Made the CHANNEL function and the CDR function process amaflags the same way. * Fixed referencing the wrong message type in cdr_prop_write(). ASTERISK-27460 Change-Id: I5eacb47586bc0b8f8ff76a19bd92d1dc38b75e8f	8 years ago
Corey Farrell	2af59ebb3a	media_index: Improve startup. This eliminates some wasteful operations in media_index startup. * Replace statically set string-fields with char[0]. * Eliminate pointless RAII_VAR's. * alloc_variant: Avoid pointless ao2_find on new info->variant. * Stop trying find_variant before alloc_variant. * process_media_file: replace ast_str with ast_asprintf. This avoids reallocation of file_id_str. Overall sounds_index.c is about 27% of Asterisk startup time when using sample configs. This patch reduces it to 20%. This is a half-fix. The real problem is that the media_index is regenerated repeatedly - 68 times in my test. Change-Id: Ia50b752f8efb356f852b05c4be495a6631af8652	8 years ago
Richard Mudgett	e97e41552e	bridge_basic.c: Update transfer diagnostic messages addendum. * Added start DTMF transfer verbose messages. * Made associated transfer messages use a similar message format. * Adjusted message verbose level as requested by initial reporter. ASTERISK-27449 Change-Id: I2045714586414b3c5ef1f3cc56c1c4af4b31f551	8 years ago
Jenkins2	3fc1490bcb	Merge "bridge_basic.c: Update transfer diagnostic messages."	8 years ago
Jenkins2	f761206f47	Merge "Add new object for VoicemailUserEntry"	8 years ago
Joshua Colp	628a0af6de	Merge "res_rtp_asterisk.c: Increase strictrtp learning timeout time."	8 years ago
Jenkins2	cc81f4eb42	Merge "pjproject: Clean up disabling of WebRTC support."	8 years ago
Niklas Larsson	9d00583164	bridge_basic.c: Update transfer diagnostic messages. * Add the channel name to diagnostic messages so you will know which channel failed to transfer. * Promoted some debug messages to verbose 4 messages. ASTERISK-27449 #close Change-Id: Idac66b7628c99379cc9269158377fd87dc97a880	8 years ago
Richard Mudgett	8536a09b86	security-events: Fix SuccessfulAuth using_password declaration. The SuccessfulAuth using_password field was declared as a pointer to a uint32_t when the field was later read as a uint32_t value. This resulted in unnecessary casts and a non-portable field value reinterpret in main/security_events.c:add_json_object(). i.e., It would work on a 32 bit architecture but not on a 64 bit big endian architecture. Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935	8 years ago
Jenkins2	890ffcd672	Merge "res_rtp_asterisk: Correct default in sample configuration file."	8 years ago
Richard Mudgett	ab63448fa6	res_rtp_asterisk.c: Increase strictrtp learning timeout time. More complicated direct media reinvite negotiations can result in longer delays before direct media flows. The strictrtp learning timeout time was too short. One log showed that the first RTP packet came in just after three seconds. * Increase the strictrtp learning timeout time from 1.5 to 5 seconds. ASTERISK-27453 Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c	8 years ago
Jenkins2	b184972d65	Merge "README-SERIOUSLY.bestpractices.txt: Convert to markdown"	8 years ago
Jenkins2	93920f1f1f	Merge "autoconf: Remove use of m4_ifblank."	8 years ago
Joshua Colp	93c2cd1383	Merge "config: Speed up config template lookup"	8 years ago
Alexander Traud	e0354bbe82	res_rtp_asterisk: Correct default in sample configuration file. With Asterisk 12 (commit `866d968`), the default of "icesupport" changed to - "yes" in the module "res_rtp_asterisk" and - "no" in the module "chan_sip". The latter was reflected in the sample configuration file for "sip.conf". The former did not make it into "rtp.conf.sample". ASTERISK-20643 Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36	8 years ago
Joshua Colp	ea1578ed7a	Merge "config: Speed up ACO & sorcery initialization"	8 years ago
Joshua Colp	1182abb0ff	Merge "res_http_post: Not all versions of gmime have GMIME_MAJOR_VERSION."	8 years ago
Sungtae Kim	0611fe581c	Add new object for VoicemailUserEntry Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's no OldMessageCount info for default. To check the OldMessageCount info, it required IMAP_STORAGE define, but this is not correct. Added OldMessageCount item as a default. ASTERISK-27456 Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7	8 years ago

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