asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	0b8a57af6d	res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. This change ensures that if no header_match option is set on an identify an error message is not output stating the option is set to an invalid value. ASTERISK-26863 Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a	8 years ago
Matt Jordan	1475604eff	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit `30f52d79d7`)	8 years ago
George Joseph	71cc3fd969	Merge "res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue"	8 years ago
Joshua Colp	c5e1e85355	Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works"	8 years ago
Joshua Colp	2ffce60844	Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item"	8 years ago
zuul	c152329932	Merge "res_pjsip_transport_websocket: Add support for IPv6."	8 years ago
zuul	2b611a8d93	Merge "chan_pjsip: Don't assume a session will have a channel."	8 years ago
Joshua Colp	578bc33f6f	Merge "chan_sip: Call not cancelled after receiving a 422 response"	8 years ago
Matt Jordan	05713c36ea	configs/samples/hep.conf.sample: Clarify how the HEP stack works This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124	8 years ago
Matt Jordan	b03b72717f	main/stasis_cache: Demote the ERROR message when removing a nonexistent item This patch demotes the ERROR message that is displayed when a nonexistent item is removed from the Stasis cache. The genesis of this demotion is due to chan_sip's realtime peers and their interaction with Asterisk's core ast_endpoint code, but ostensibly it could happen from other channel drivers as well. Since Mark Michelson already did an excellent job of explaining on this issue, it is quoted here for posterity: "Internally, when a realtime peer is retrieved, Asterisk creates an ast_endpoint structure. When that peer is destroyed, the ast_endpoint is destroyed as well. Part of the destruction of the ast_endpoint involves clearing the Stasis cache of all information about that endpoint. The problem here is that the act of creating the ast_endpoint is not enough to actually put any information in the Stasis cache. Instead, something has to happen, such as a state change, in order for the Stasis cache to have any information about that endpoint. When a device registers, chan_sip creates an ast_endpoint structure, processes the REGISTER, and then destroys the ast_endpoint. When the ast_endpoint is destroyed, there is nothing to destroy in the Stasis cache, so an error message is emitted. When you use rtcachefriends, ast_endpoint structures persist for the lifetime of the module and so you do not see this error message." ASTERISK-25237 #close Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70	8 years ago
Matt Jordan	2d7e68c075	res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue Tabs > spaces. Always. Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1	8 years ago
Joshua Colp	12460b05c1	chan_pjsip: Don't assume a session will have a channel. When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01	8 years ago
George Joseph	d1ef127084	pjproject_bundled: Reduce the need for rebuilds Bundled pjproject should now only rebuild if one of the menuselect "Compiler Flags" options changes. Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43	8 years ago
Joshua Colp	018e01543d	Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"	8 years ago
zuul	87aaaef8bb	Merge "res_musiconhold: moh general section is a class and issues warning"	8 years ago
Joshua Colp	9250ddb8f2	Merge "media_cache: Prefer ast_file_is_readable() over access()"	8 years ago
Daniel Journo	36fed72614	pjsip/cli_commands: pjsip show channelstats shows wrong codec * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01	8 years ago
Joshua Colp	6113bb470b	Merge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"	8 years ago
Daniel Journo	b14724adb3	res_musiconhold: moh general section is a class and issues warning * res_musiconhold.c: Ensure the general section is not treated as a moh class. ASTERISK-26353 #close Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d	8 years ago
Sean Bright	35cfd2c0cc	media_cache: Prefer ast_file_is_readable() over access() Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def	8 years ago
Sean Bright	bc2c66b594	pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel Set a variable on the channel that indicates which attempt number we are currently performing to allow for attempt-specific behavior. ASTERISK-26568 #close Reported by: Roman Shubovich Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89	8 years ago
Joshua Colp	4e3b0cedba	res_pjsip_transport_websocket: Add support for IPv6. This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647	8 years ago
Daniel Journo	60998371e3	app_voicemail: Cannot set fromstring on a per-mailbox basis * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	8 years ago
zuul	ff0569740a	Merge "res_http_websocket: Fix faulty read logic."	8 years ago
zuul	9cd1a754a9	Merge "pbx_spool: Gracefully handle long lines in call files"	8 years ago
Mark Michelson	5d0371d743	res_http_websocket: Fix faulty read logic. When doing some WebRTC testing, I found that the websocket would disconnect whenever I attempted to place a call into Asterisk. After looking into it, I pinpointed the problem to be due to the iostreams change being merged in. Under certain circumstances, a call to ast_iostream_read() can return a negative value. However, in this circumstance, the websocket code was treating this negative return as if it were a partial read from the websocket. The expected length would get adjusted by this negative value, resulting in the expected length being too large. This patch simply adds an if check to be sure that we are only updating the expected length of a read when the return from a read is positive. ASTERISK-26842 #close Reported by Mark Michelson Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab	8 years ago
Jean Aunis	d51ca4b406	chan_sip: Call not cancelled after receiving a 422 response When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099	8 years ago
Daniel Journo	272259a2c6	Saynumber is trying to get "and" from "digits/" subfolder * say.c Changed 'digits/and' to 'vm-and' for en_GB ASTERISK-26598 #close Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe	8 years ago
Sean Bright	5a74abc53b	pbx_spool: Gracefully handle long lines in call files Per the linked issue, we aren't checking the buffer filled by fgets() to determine if it contains a newline, so we will fail to correctly parse the trailing portion of a long line. This patch increases the buffer size from 256 to 1024, and skips any line that exceeds that length, logging a warning in the process. ASTERISK-17067 #close Reported by: Dave Olszewski Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0	8 years ago
Richard Mudgett	c9296b23d1	core: Cleanup ast_get_hint() usage. * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b	8 years ago
Joshua Colp	e9b2360d17	Merge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS."	8 years ago
Joshua Colp	fb11f038a3	Merge "stream: Unit tests for stream read and tweaks framework"	8 years ago
Joshua Colp	9b0e142a6a	Merge "res_config_pgsql: Make 'require' return consistent with other backends"	8 years ago
Jørgen H	7922f26cb0	res_pjsip WebRTC/websockets: Fix usage of WS vs WSS. According to the RFC[1] WSS should only be used in the Via header for secure Websockets. * Use WSS in Via for secure transport. * Only register one transport with the WS name because it would be ambiguous. Outgoing requests may try to find the transport by name and pjproject only finds the first one registered. This may mess up unsecure websockets but the impact should be minimal. Firefox and Chrome do not support anything other than secure websockets anymore. * Added and updated some debug messages concerning websockets. * security_events.c: Relax case restriction when determining security transport type. * The res_pjsip_nat module has been updated to not touch the transport on Websocket originating messages. [1] https://tools.ietf.org/html/rfc7118 ASTERISK-26796 #close Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12	8 years ago
George Joseph	0560c32375	stream: Unit tests for stream read and tweaks framework * Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2	8 years ago
Sean Bright	1dacf317f3	res_config_pgsql: Make 'require' return consistent with other backends res_config_pgsql should match the behavior of other realtime backend drivers so that queue_log can disable adaptive logging. ASTERISK-25628 #close Reported by: Dmitry Wagin Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372	8 years ago
Mark Michelson	9c55a71798	SDP: Add initial SDP state machine. This introduces and documents the various states in the state machine. This also introduces API functions that induce state changes, and places TODO comments telling what needs to be done in addition to what is already there. Those TODOs will be replaced with real code in upcoming changes. Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed	8 years ago
Joshua Colp	26bf1846e2	Merge "media_cache: Mark cache entry stale if cache file is removed"	8 years ago
Joshua Colp	10d12b277c	Merge "res_config_pgsql: Release table locks where appropriate"	8 years ago
Joshua Colp	063af910eb	Merge "res_pjsip_outbound_registration: Subscribe to network change events"	8 years ago
Joshua Colp	6d3c1a4a21	Merge "build: Warn if asterisk is installed in both 32 and 64 bit sys dirs"	8 years ago
Joshua Colp	5e5aff04ec	Merge "res_pjsip_pubsub: Remove unneeded endpoint unref"	8 years ago
zuul	f41ace9042	Merge "bridge_native_rtp: Handle case where channel joins already suspended."	8 years ago
Sean Bright	60e9e4fcc0	media_cache: Mark cache entry stale if cache file is removed In the event that a cache file is removed out from under us, we should treat the cache entry as stale and force a refresh. ASTERISK-26774 #close Reported by: Igor Gamayunov Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52	8 years ago
Joshua Colp	0986998f2f	Merge "config: Improve documentation and behavior of outbound_proxy option."	8 years ago
Joshua Colp	9c714b03f9	Merge "res_pjsip: Fix crash when contact has no status"	8 years ago
Sean Bright	e5b44c26b4	res_config_pgsql: Release table locks where appropriate The find_table() functions NULL or a locked table pointer. We are not consistently calling release_table() in failure paths. Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544	8 years ago
Tzafrir Cohen	6ebdcfe27d	pjsip.conf.sample: user_agent: not a specific version Use the description of useragent from sip.conf here. ASTERISK-26825 #close Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755	8 years ago
George Joseph	fb68db87b1	res_pjsip_pubsub: Remove unneeded endpoint unref When a subscription was being recreated and the endpoint wasn't found, we were trying to unref the endpoint. This was causing FRACKs. Removed the unref. ASTERISK-26823 #close Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164	8 years ago
Jørgen H	ee0a123f43	res_pjsip: Fix crash when contact has no status This change fixes an assumption in res_pjsip that a contact will always have a status. There is a race condition where this is not true and would crash. The status will now be unknown when this situation occurs. ASTERISK-26623 #close Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5	8 years ago

1 2 3 4 5 ...

29150 Commits (0b8a57af6d62c19e3ed4b37b06a55484c3a04092) All Branches Search

29150 Commits (0b8a57af6d62c19e3ed4b37b06a55484c3a04092)

All Branches