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Author	SHA1	Message	Date
zuul	d9ed4e3153	Merge "pjproject_bundled: Various fixes discovered during testing of OSes"	10 years ago
zuul	c339d4c6ed	Merge "pjsip: Added "reg_server" to contacts."	10 years ago
Joshua Colp	96de883676	Merge "configs/basic-pbx/asterisk.conf: contains incorrect path separator"	10 years ago
George Joseph	e61716b774	pjproject_bundled: Various fixes discovered during testing of OSes ... For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c	10 years ago
zuul	9c4c4a7d0d	Merge "res_pjsip_exten_state: Create PUBLISH messages."	10 years ago
Alexei Gradinari	2b1edee772	pjsip: Added "reg_server" to contacts. ... If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8	10 years ago
Diederik de Groot	bf13b59062	configs/basic-pbx/asterisk.conf: contains incorrect path separator ... Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004	10 years ago
Joshua Colp	adf32d1191	Merge "pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE"	10 years ago
Richard Mudgett	2c46063d54	res_pjsip_exten_state: Create PUBLISH messages. ... Create PUBLISH messages to update a third party when an extension state changes because of either a device or presence state change. A configuration example: [exten-state-publisher] type=outbound-publish server_uri=sip:instance1@172.16.10.2 event=presence ; Optional regex for context filtering, if specified only extension state ; for contexts matching the regex will cause a PUBLISH to be sent. @context=^users ; Optional regex for extension filtering, if specified only extension ; state for extensions matching the regex will cause a PUBLISH to be sent. @exten=^[0-9]* ; Required body type for the PUBLISH message. ; ; Supported values are: ; application/pidf+xml ; application/xpidf+xml ; application/cpim-pidf+xml ; application/dialog-info+xml (Planned support but not yet) @body=application/pidf+xml The '@' extended variables are used because the implementation can't extend the outbound publish type as it is provided by the outbound publish module. That means you either have to use extended variables, or implement some sort of custom extended variable thing in the outbound publish module. Another option would be to refactor that stuff to have an option which specifies the use of an alternate implementation's configuration and then have that passed to the implementation. JColp opted for the extended variables method originally. ASTERISK-25972 #close Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca	10 years ago
Joshua Colp	bc19d9a2b0	Merge "res_pjsip_exten_state: Check if body generator is available."	10 years ago
Joshua Colp	d57847a7c7	Merge "res_pjsip_pubsub.c: Fix body generator registration race."	10 years ago
zuul	ce3687011f	Merge "res_pjsip: Start body generator users after suppliers."	10 years ago
Joshua Colp	1e41d14822	Merge "chan_sip: Make autocreated peers send PeerStatus events"	10 years ago
zuul	e4b086939d	Merge "res_pjsip_pubsub.c: Add useful information to some messages."	10 years ago
zuul	9692f8543e	Merge "res_pjsip_pubsub.h: Fix doxygen association."	10 years ago
zuul	c8f53bc4e9	Merge "res_pjsip_outbound_publish.c: Remove redundant flag check."	10 years ago
zuul	980b772265	Merge "res_pjsip: Add ability to identify by Authorization username"	10 years ago
zuul	d53d494f0b	Merge "app_chanspy: reduce audio loss on the spying channel."	10 years ago
Richard Mudgett	0b5292525c	res_pjsip_exten_state: Check if body generator is available. ... When starting the extension state publishers, check if the requested message body generator is available. If not available give error message and skip starting that publisher. * res_pjsip_pubsub.c: Create new API if type/subtype generator registered. * res_pjsip_exten_state.c: Use new body generator API for validation. ASTERISK-25922 Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c	10 years ago
Richard Mudgett	369182d084	res_pjsip: Start body generator users after suppliers. ... Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb	10 years ago
Richard Mudgett	3af83ea2fb	res_pjsip_pubsub.c: Add useful information to some messages. ... Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a	10 years ago
Richard Mudgett	8e1b663b87	res_pjsip_pubsub.c: Fix body generator registration race. ... Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67	10 years ago
George Joseph	30415944a8	pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE ... When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, the input uri string will become corrupted if it contains escape sequences. It's not possible to automatically strdup or strdupa the input string because the output uri pj_str_t's will have pointers to chunks of the input string. Getting around this would require more memory management code and wouldn't be worth the savings of doing the unescape in place. ASTERISK-25970 #close Reported-by: Dmitriy Serov Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88	10 years ago
Richard Mudgett	906ea2c43f	res_pjsip_pubsub.h: Fix doxygen association. ... Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632	10 years ago
Richard Mudgett	76ea4cfaae	res_pjsip_outbound_publish.c: Remove redundant flag check. ... Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353	10 years ago
zuul	057ed94048	Merge "res_pjsip_exten_state: Add config support for exten state publishers."	10 years ago
zuul	9309a96ae1	Merge "func_odbc: Check connection status before executing queries."	10 years ago
George Joseph	4ebf9a938d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	10 years ago
Joshua Colp	f324801763	Merge "config: Fix ast_config_text_file_save2 writability check for missing files"	10 years ago
Mark Michelson	2b150f0b80	func_odbc: Check connection status before executing queries. ... A recent change to func_odbc made it so that a single connection was maintained per DSN. The problem was that the code was optimistic about the health of the connection after initially opening it and did nothing to re-connect in case the connection had died. This change adds a check before executing a query to ensure that the connection to the database is still up and running. ASTERISK-25963 #close Reported by Ross Beer Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d	10 years ago
Joshua Colp	d1b9b96456	Merge "res_pjsip: disable multi domain to improve realtime performace"	10 years ago
zuul	9d57416315	Merge "res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)"	10 years ago
Alexei Gradinari	860b135c88	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	10 years ago
Jean Aunis	7281770710	app_chanspy: reduce audio loss on the spying channel. ... ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when queues grow too large or when read and write queues go out of sync. Now these flags are set conditionally: - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not be set on the audiohook ASTERISK-25866 Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd	10 years ago
Joshua Colp	81ea80b74c	res_pjsip_exten_state: Add config support for exten state publishers. ... This change adds the ability to configure outbound publishing of extension state. Right now stuff is merely set up to store the configuration and to register a global extension state callback. The act of constructing the body and sending is not yet complete. Configurable elements right now are a regex for filtering the context, a regex for filtering the extension, and the body type to publish. ASTERISK-25922 #close Change-Id: Ia7e630136dfc355073c1cadff8ad394a08523d78	10 years ago
Joshua Colp	c480159045	chan_sip: Give more time for TCP/TLS threads to stop. ... The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6	10 years ago
Joshua Colp	8ae69cffef	app_queue: Fix crash when unloading module. ... When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b	10 years ago
Joshua Colp	09d588dc2f	Merge changes from topic 'system_stress_patches' ... * changes: test_message.c: Wait longer in case dialplan also processes the test message. Manager: Short circuit AMI message processing. manager.c: Eliminate most RAII_VAR usage.	10 years ago
zuul	0f29785101	Merge "manager_channels.c: Fix allocation failure crash."	10 years ago
zuul	811e24f595	Merge "Bridge system: Fix memory leaks and double frees on impart failure."	10 years ago
zuul	807a765cfb	Merge "bridge_softmix.c: Fix crash if channel fails to join mixing tech."	10 years ago
Joshua Colp	456600a641	Merge "app_queue: queue members can receive multiple calls"	10 years ago
George Joseph	284bb814ac	config: Fix ast_config_text_file_save2 writability check for missing files ... A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80	10 years ago
DarkS	f99ec857c8	Fix case sensitive actions in AMI QueueSummary and QueueStatus ... ASTERISK-25954 #close Reported by: Javier Acosta Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 (cherry picked from commit c0688a6398)	10 years ago
Kevin Harwell	30ab21d5fa	app_queue: queue members can receive multiple calls ... It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48	10 years ago
George Joseph	99fcf2a791	res_agi: Prevent run_agi from eating frames it shouldn't ... The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645	10 years ago
Joshua Colp	7f8d83fef4	Merge "func_odbc: Use one connection per DSN."	10 years ago
zuul	0bd3444654	Merge "Remove reference to non-existent sip.conf option"	10 years ago
zuul	ac50fdecdb	Merge "res_stasis: Handle re-enter stasis bridge with swap channel."	10 years ago
zuul	1df086f821	Merge "bridge: Hold off more than one imparting channel at a time."	10 years ago