28910 Commits (812f5b51cb56a36668decc6dfc83adeca185429e)

 

All Branches   

Search

Author	SHA1	Message	Date
Joshua Colp	701b753a0b	Merge "chan_iax2: Reload of iax peer results in loss of host address/port" into 13	8 years ago
zuul	9a5c028107	Merge "app_queue: Handle the caller being redirected out of a queue bridge" into 13	8 years ago
zuul	05ab662f70	Merge "pbx.c: Fix crash from malformed exten pattern." into 13	8 years ago
zuul	bcc566b77e	Merge "res/res_pjsip_refer: call xfer w/o extension" into 13	8 years ago
Richard Mudgett	adad6020be	autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. ... Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592	8 years ago
zuul	481ed5c8f1	Merge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match." into 13	8 years ago
zuul	aa5d108caa	Merge "configure: Don't use the progress bar with curl when downloading to stdout" into 13	8 years ago
Mark Michelson	7bc69753bc	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Sean Bright	163e9e53dc	app_queue: Handle the caller being redirected out of a queue bridge ... A caller can leave the Queue() application after being bridged with a member in a few ways: * Caller or member hangup * Caller is transferred somewhere else (blind or atx) * Caller is externally redirected elsewhere The first 2 scenarios are currently handled by subscribing to stasis messages, but the 3rd is not explicitly covered. If a caller is redirected away from the Queue() application, the member who was last bridged with that caller will remain in an "In use" state until the caller hangs up. This patch adds handling of the caller leaving the queue via redirection. We monitor the caller-member bridge, and if the caller is the one that leaves, we treat it the same as we would a caller hangup. ASTERISK-26400 #close Reported by: Etienne Lessard Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334	8 years ago
Joshua Colp	7612601964	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
Joshua Colp	947f1ebf86	Merge "chan_pjsip: Don't assume a session will have a channel." into 13	8 years ago
Joshua Colp	0f87808414	Merge "menuselect: Add a new 'options' support type" into 13	8 years ago
Joshua Colp	475dfdb88c	Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works" into 13	8 years ago
Joshua Colp	47ccdd45ca	Merge "res_pjsip_endpoint_identifier_ip: Add an option to match requests by header" into 13	8 years ago
Joshua Colp	45d5c04fd0	Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item" into 13	8 years ago
Torrey Searle	48447313b6	res/res_pjsip_refer: call xfer w/o extension ... When transfering to a URI without an extension, ensure that the s extension of the dialplan is entered ASTERISK-26869 #close Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525	8 years ago
zuul	683321cba9	Merge "funcs/func_devstate: Remove new line in Device field of during module load" into 13	8 years ago
Joshua Colp	86c69ccdf7	Merge "res_pjsip_transport_websocket: Add support for IPv6." into 13	8 years ago
Richard Mudgett	9fd9b39e8b	pbx.c: Fix crash from malformed exten pattern. ... Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a	8 years ago
Richard Begg	5389666d6f	chan_iax2: Reload of iax peer results in loss of host address/port ... When using a non-dynamic peer address, build_peer() invalidates the peer address structure by setting the address family to unspecified. However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() will not amend the peer address if the cache is still valid, resulting in peer connectivity failures. To fix this, we call ast_dnsmgr_refresh() instead. ASTERISK-26865 Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082	8 years ago
Matt Jordan	658d59c683	configure: Don't use the progress bar with curl when downloading to stdout ... In some scenarios, such as when there may not be a terminal (such as inside a Docker container), curl will apparently direct the progress bar to stdout. This can cause extra data to be appended to a file curl'd down to stdout, resulting in md5 verification failures. This patch removes the progress bar, and tells curl to download the file silently. ASTERISK-26872 #close Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c	8 years ago
Matt Jordan	b3c2c996f1	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
Matt Jordan	51985565ef	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
zuul	071e6d971c	Merge "chan_sip: Call not cancelled after receiving a 422 response" into 13	8 years ago
Matt Jordan	f9b791debe	funcs/func_devstate: Remove new line in Device field of during module load ... During module loading of func_devstate, Asterisk emits the current device state of all Custom device states currently stored in the AstDB. This was erroneously including a new line character ('\n') to the end of the device state, causing two new lines to be emitted in DeviceStateChange AMI events. Note that this only happened for those device state changes that occurred during startup. Regular device state changes for Custom device states are handled elsewhere, and did not have the newline. ASTERISK-26643 #close Reported by: Roman Bedros Tested by: Matt Jordan patches: ami_devstate.diff uploaded by Roman Bedros (License 6842) Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93	8 years ago
Matt Jordan	216e28aa95	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
Joshua Colp	c8d1b915d7	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	6d1eb880c2	menuselect: Add a new 'options' support type ... The Binaural Rendering patches in the master branch required menuselect to be updated with a new support type called 'option'. This allows binaural rendering to be turned on or off when bridge_softmix is built. This patch backports the 'option' functionality to the 13 and 14 branches. Here's what it looks like in menuselect: [*] bridge_simple [*] bridge_softmix --- Module Options --- [ ] binaural_rendering_in_bridge_softmix To create an option for a module, you can create (or update) the menuselect-tree xml snippet in the directory where the module resides and add a member element with an 'option' support_level. Example (abbreviated) from bridges/bridges.xml: <member name="binaural_rendering_in_bridge_softmix" displayname="Enable binaural rendering in bridge_softmix" remove_on_change="bridges/bridge_softmix.o bridges/bridge_softmix.so"> <support_level>option</support_level> <depend>bridge_softmix</depend> <depend>fftw3</depend> <defaultenabled>no</defaultenabled> </member> The 'name' will be added or removed from the MENUSELECT_<dir> make variable following the standard module "missing means yes" rules. Example (abbreviated) from bridges/Makefile: ifeq ($(findstring binaural_rendering,$(MENUSELECT_BRIDGES)),) bridge_softmix.o: _ASTCFLAGS+=-DBINAURAL_RENDERING bridge_softmix.so: LIBS+=$(FFTW3_LIB) endif Change-Id: I66d23755ed6e81f8d439cad410f2ffa7c30f25ad	8 years ago
George Joseph	523de8eb8e	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
Jean Aunis	d3ef833b3b	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
Joshua Colp	8386a38e06	Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec" into 13	8 years ago
Daniel Journo	67c989ce78	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	2a85888262	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	bc6eeab822	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
Joshua Colp	a860bf63e1	Merge "pbx_spool: Gracefully handle long lines in call files" into 13	8 years ago
Daniel Journo	d9972423d1	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	77901a58ca	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	4271c700f7	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	75ebd8f0d2	Merge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS." into 13	8 years ago
zuul	65a025e9fa	Merge "chan_sip: Allow DTLS to be disabled when reloading." into 13	8 years ago
Jørgen H	e510595c86	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
Sean Bright	76971d4c4a	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
zuul	848e211e1c	Merge "res_config_pgsql: Release table locks where appropriate" into 13	8 years ago
zuul	c36cab8468	Merge "res_pjsip_outbound_registration: Subscribe to network change events" into 13	8 years ago
zuul	4bff6bc144	Merge "build: Warn if asterisk is installed in both 32 and 64 bit sys dirs" into 13	8 years ago
zuul	024e724aca	Merge "res_pjsip_pubsub: Remove unneeded endpoint unref" into 13	8 years ago
zuul	acb906fa2f	Merge "bridge_native_rtp: Handle case where channel joins already suspended." into 13	8 years ago
zuul	59a00786e8	Merge "config: Improve documentation and behavior of outbound_proxy option." into 13	8 years ago
zuul	bf3cb76452	Merge "res_pjsip: Fix crash when contact has no status" into 13	8 years ago
Sean Bright	fa8f6c2fc4	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