29136 Commits (12460b05c1abd37e412e155e47a6e349a2df3338)

 

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Author	SHA1	Message	Date
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
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
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
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
George Joseph	22242fef5d	res_pjsip_outbound_registration: Subscribe to network change events ... Outbound registration now subscribes to network change events published by res_stun_monitor and refreshes all registrations when an event happens. The 'pjsip send (un)register' CLI commands were updated to accept '*all' as an argument to operate on all registrations. The 'PJSIP(Un)Register' AMI commands were also updated to accept '*all'. ASTERISK-26808 #close Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25	8 years ago
George Joseph	4692a32ed7	build: Warn if asterisk is installed in both 32 and 64 bit sys dirs ... ... and clean them both up on uninstall. We've fixed the issue where 'make install' was installing to /usr/lib on 64-bit systems that use /usr/lib64. Now we need to clean up the remnants in /usr/lib. * 'make install' now prints a warning if DESTDIR/ASTLIBDIR contains 'lib64' and libasterisk* shared libraries or modules are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. * 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. ASTERISK-26705 Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f	8 years ago
Joshua Colp	ff2b4308d1	bridge_native_rtp: Handle case where channel joins already suspended. ... The bridge_native_rtp module did not properly handle the case where a smart bridge operation occurs while a channel is suspended. In this scenario the module would incorrectly set up local or remote RTP bridging despite the media having to flow through Asterisk. The remote endpoint would see two media streams and experience wonky audio. The module has been changed so that it ensures both channels are not suspended when performing the native RTP bridging and this requirement has been documented in the bridge technology. ASTERISK-26781 Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c	8 years ago
George Joseph	df22d297a6	Merge "channel: Add ast_read_stream function for reading frames from all streams."	8 years ago
zuul	00d1c7ddd2	Merge "Binaural synthesis (confbridge): DTMF conference management."	8 years ago
frahaase	5b1796f59d	Binaural synthesis (confbridge): DTMF conference management. ... DTMF configuration options for the binaural softmix bridge: toggle binaural rendering (per channel). ASTERISK-26292 Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8	8 years ago
Joshua Colp	2046743938	config: Improve documentation and behavior of outbound_proxy option. ... This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593	8 years ago
Joshua Colp	e4ba07f1a6	Merge "pjproject_bundled: Update for pjproject 2.6"	8 years ago
Joshua Colp	6ac33bfe3e	Merge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."	8 years ago
zuul	f943ead12e	Merge "build: Execute ldconfig to build cache. (take two)"	8 years ago
zuul	461577b23b	Merge "channel: Add support for writing to a specific stream."	8 years ago
Joshua Colp	c07c6714f2	channel: Add ast_read_stream function for reading frames from all streams. ... This change introduces an ast_read_stream function and callback in the channel technology which allows reading frames from all streams and not just the default streams. The stream number has also been added to frames. This is to allow the case where frames are queued onto the channel instead of being read directly from the driver. This change does impose a restriction on reading though: a chain of frames can only contain frames from the same stream. ASTERISK-26816 Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce	8 years ago
George Joseph	a537dae6d0	pjproject_bundled: Update for pjproject 2.6 ... * Removed all 2.5.5 functional patches. * Updated usages of pj_release_pool to be "safe". * Updated configure options to disable webrtc. * Updated config_site.h to disable webrtc in pjmedia. * Added Richard Mudgett's recent resolver patches. Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7	8 years ago
George Joseph	b0067bcf2c	build: Execute ldconfig to build cache. (take two) ... On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. To make matters worse, options were being passed to ldconfig on both Linux and FreeBSD that actually prevented the rebuild of the cache. * Fedora has a /usr/share/config.site that automatically tells autoconf to use /usr/lib64 but CentOS does not. This logic was copied to configure.ac and modified so systems like Ubuntu, which still use /usr/lib for 64-bit systems, aren't affected. Now that we have them in the correct directory... In order for the system loader to find libasteriskssl and libasteriskpj, one of 3 things has to happen... - The linker cache must be rebuilt including the directory where the libasterisk* libraries were installed. Only root can rebuild the cache. This was busted. - We have to link the asterisk binary with an rpath pointing to the directrory where the libasterisk* libraries were installed. This makes things very complicated and will happen over the collective dead bodies of everyone who's had to package a distribution with an rpath. - Finally, you can start asterisk with LD_LIBRARY_PATH set to the directrory where the libasterisk* libraries were installed. There are no other options. So... * The invokation of ldconfig has been moved from main/Makefile to ASTTOPDIR/Makefile, the options have been removed, and DESTDIR/ASTLIBDIR appended. If you aren't root, you will be warned after the "Asterisk Installation Compete" banner that you must re-run 'make install' as root, manually run 'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with LD_LIBRARY_PATH. ASTERISK-26705 Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982	8 years ago