4889 Commits (9d5797616c840e89fdb12b1b49e59f1266b51a68)

Author	SHA1	Message	Date
Joshua Colp	7a837ece97	Merge "Add new AMI action for app_voicemail"	7 years ago
pchero	daa3a3009a	Add new AMI action for app_voicemail ... Currently, to figure out specified voicemail's status, there's only one way to do it, which is use a VoicemailUserEntry AMI message. But it consumed it too much resource(it check everything). So, added new AMI action. ASTERISK-27470 Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7	7 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
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
Corey Farrell	58fa3885cc	Loader: Remove unneeded load_pri declarations. ... Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT. Change-Id: I0123258eafce324249433a69df15a85cc16e509f	8 years ago
Corey Farrell	2fab3aacd6	app_minivm: Fix possible uninitialized return value. ... Declare 'res' initialized to -1 to deal with earlier error paths that could cause 'res' to be returned uninitialized. Change-Id: I8ac2a5755bf4174d89ef893e924c940f702b104e	8 years ago
George Joseph	f512707362	app_record: Don't set RECORD_STATUS chan var until file is closed ... We've been calling pbx_builtin_setvar_helper to set the RECORD_STATUS variable before actually closing the recorded file. If a client is watching VarSet events and tries to do something with the file when a RECORD_STATUS event is seen, they might attempt to do so while the file it's still open. We now delay calling pbx_builtin_setvar_helper until after we close the file. ASTERISK-27423 Change-Id: I7fe9de99953e46b4bafa2b38cf151fe8f6488254	8 years ago
Richard Mudgett	ee08f10d06	Fix ast_(v)asprintf() malloc failure usage conditions. ... When (v)asprintf() fails, the state of the allocated buffer is undefined. The library had better not leave an allocated buffer as a result or no one will know to free it. The most likely way it can return failure is for an allocation failure. If the printf conversion fails then you actually have a threading problem which is much worse because another thread modified the parameter values. * Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL on failure. That is much more useful than either an uninitialized pointer or a pointer that has already been freed. Many uses won't have to check for failure to ensure that the buffer won't be double freed or prevent an attempt to free an uninitialized pointer. * stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by ast_asprintf(). * ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to the wrong thing which is now not needed even if assigning to the right thing. Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23	8 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	8 years ago
Joshua Colp	92b3ed4d32	Merge "Modules: Fix issues with CLI completion."	8 years ago
Corey Farrell	e82b921c35	Modules: Fix issues with CLI completion. ... * Stop using ast_module_helper to check if a module is loaded, use ast_module_check instead (app_confbridge and app_meetme). * Stop ast_module_helper from listing reload classes when needsreload was not requested. ASTERISK-27378 Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239	8 years ago
Igor Goncharovskiy	9bad4c74cc	app_agent_spool: Fix typo in dtmf features usage desctiption ... Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function instead of correct 'dtmf_features' ASTERISK-27377 #close Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930	8 years ago
Richard Mudgett	26607e4e3b	app_voicemail.c: Fix compiler warning with IMAP build. ... ASTERISK-27181 Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17	8 years ago
Richard Mudgett	488f98310f	app_agent_pool.c: Fix online documentation typo. ... Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429	8 years ago
Corey Farrell	955a891a84	app_macro deprecation. ... * Mark the module deprecated. * Disable the module by default. * Produce a warning the first time a macro is used. * Note deprecation related options in app_dial and app_queue. ASTERISK-27350 Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc	8 years ago
Joshua Colp	0bedd436dc	Merge "app_originate: Set ORIGINATE_STATUS correctly on failure"	8 years ago
Nathan Bruning	be7da57546	app_queue.c: clear moh field in init_queue ... ASTERISK-27301 #close Change-Id: Ic31361f34e2de3b6470e68fc37205a7711082eba	8 years ago
Sean Bright	3ad7d2f36c	app_originate: Set ORIGINATE_STATUS correctly on failure ... We were ignoring the return value from ast_pbx_outgoing_exten() and ast_pbx_outgoing_app() which could fail before setting the reason code. This resulted in failures being reported as success. ASTERISK-25266 #close Reported by: Allen Ford Change-Id: Idf16237b7e41b527d2c69c865829128686beeb3b	8 years ago
Richard Mudgett	b2dbfe23ef	app_queue.c: Fix announcements when announce-to-first-user not enabled. ... The previous patch for ASTERISK-27216 made it so you wouldn't get any position or periodic announcements unless you had announce-to-first-user enabled. The announce-to-first-user feature was added by ASTERISK_21782 as a result of the patch which introduced the redundant announcements that ASTERISK-27216 removes. * By noting that the makeannouncement variable is used to suppresses the first user announcement, we set its initial value to the announce-to-first-user enable setting. ASTERISK-27216 Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a	8 years ago
Joshua Colp	2a09b3b76a	Merge "app_queue: Only do announcement logic between ringing cycles"	8 years ago
Joshua Colp	0133d29c83	Merge "build: A few gcc 7 error fixes"	8 years ago
George Joseph	4275ca16a1	build: A few gcc 7 error fixes ... Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec	8 years ago
StefanEng86	c3c73b3511	app_queue: Only do announcement logic between ringing cycles ... This patch reverts the change by patch 2263 from old reviewboard. Note that reverting that 2263-patch still preserves the behaviour that the commit log of the 2263-patch claimed to add. The reason for this is: The function wait_for_answer is only called from try_calling which in turn is only called from the main for loop in queue_exec, and earlier in that loop we already check the things that's removed by this patch. There's no need to check those things twice each loop iteration, and I think the proper place to check it is before each ringing cycle. By checking it in wait_for_answer, you allow the issue explained in the jira - that the head caller hears announcements while the agents' sip phones are actively ringing. Reported-by: Stefan Engström Tested-by: Stefan Engström ASTERISK-27216 #close Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0	8 years ago
Sean Bright	0fad11f21c	app_stream_echo: Don't echo declined streams ... Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic Edition after accepting the audio request but declining the video one. Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c	8 years ago
Joshua Colp	6fd3db51e8	app_confbridge: Only create a channel that records audio. ... This change makes it so that the conference recorder channel that is created only contains audio formats and an audio stream. This is because the underlying application used by ConfBridge to record, MixMonitor, only allows recording audio. Having additional streams (and in particular a video stream) can result in clients needlessly renegotiating to add a video stream that will never receive video. Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0	8 years ago
Sean Bright	2b3f903e6f	app_waitforsilence: Cleanup & don't treat missing frames as 'noise' ... * WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9	8 years ago
Sean Bright	1bdbefbe76	app_directory: Handle a NULL mailbox without crashing ... ASTERISK-27241 #close Reported by: David Moore Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6	8 years ago
Corey Farrell	1bf3dfffd7	AST-2017-006: Fix app_minivm application MinivmNotify command injection ... An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	8 years ago
Sean Bright	06cc5ae9ff	confbridge: Handle user hangup during name recording ... This prevents orphaned CBAnn channels from getting stuck in the bridge. ASTERISK-26994 #close Reported by: James Terhune Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457	8 years ago
Jenkins2	f47733cea0	Merge "app_record: Resolve some absolute vs. relative filename bugs"	8 years ago
Sean Bright	da13cdb9e7	voicemail: Fix various abuses of mkstemp ... mkstemp() returns a unique filename, but appending an extension to that filename does not guarantee uniqueness. Instead, use mkdtemp() and we can put whatever extension we want on the files that we create inside the directory. In the case of app_minivm, we also now properly clean up any temporary files that we create. ASTERISK-20858 #close Reported by: Walter Doekes Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43	8 years ago
Sean Bright	43670e471f	app_record: Resolve some absolute vs. relative filename bugs ... If the Record() application is called with a relative filename that includes directories, we were not properly creating the intermediate directories and Record() would fail. Secondarily, updated the documentation for RECORDED_FILE to mention that it does not include a filename extension. Finally, rewrote the '%d' functionality to be a bit more straight forward and less noisy. ASTERISK-16777 #close Reported by: klaus3000 Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2	8 years ago
Jenkins2	179524c255	Merge "app_queue: Evaluate realtime queues when running dialplan functions"	8 years ago
Sean Bright	02f95d290f	app_queue: Evaluate realtime queues when running dialplan functions ... ASTERISK-19103 #close Reported by: Jim Van Meggelen Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b	8 years ago
Sean Bright	b1097be134	app_voicemail: Honor escape digits in "greeting only" mode ... ASTERISK-21241 #close Reported by: Eelco Brolman Patches: Patch uploaded by Eelco Brolman (License 6442) Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe	8 years ago
Joshua Colp	c7edd0b493	Merge "app_confbridge: Document sfu video_mode value."	8 years ago
Jenkins2	e8cf4ab833	Merge "confbridge.h: Fix doxygen comments."	8 years ago
Richard Mudgett	5bbf7b2aad	app_confbridge: Document sfu video_mode value. ... Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204	8 years ago
Richard Mudgett	f96536b1ea	confbridge.h: Fix doxygen comments. ... Change-Id: I16133166a85fdb557c66ffcbfe8128d0b4725b0e	8 years ago
Sungtae Kim	22af5e3784	app_queue: Fix initial hold time queue statistic ... Fixed to use correct initial value and fixed to use the correct queue info to check the first value. ASTERISK-27204 Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73	8 years ago
Jenkins2	1fde9dc7a5	Merge "app_privacy: remove unused header asterisk/image.h"	8 years ago
Corey Farrell	4b03eb5c38	Fix compile error for old versions of GCC. ... Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	8 years ago
Corey Farrell	148cf2e0f7	app_privacy: remove unused header asterisk/image.h ... Change-Id: I56ed530633a642633b18383821069e806c92ae82	8 years ago
Corey Farrell	58d032112b	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	8 years ago
Sean Bright	3f98488279	app_queue: Add announce-position-only-up option ... Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6	8 years ago
Joshua Colp	b610295b62	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."	8 years ago
Richard Mudgett	922930753c	app_voicemail.c: Allow mailbox entry on authentication retry prompt. ... The following testsuite voicemail tests were failing to re-enter the mailbox after the first login attempt. tests/apps/voicemail/authenticate_invalid_mailbox tests/apps/voicemail/authenticate_invalid_password The tests were noting the start of the vm-incorrect-mailbox prompt and immediately sending the mailbox for the next login attempt. Since the invalid message playback had to complete before the digits were recognized, the test passed for the wrong reason and added approximately 20 seconds to the test times. * Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox digits like the initial vm-login prompt so the tests are able to enter the intended mailbox. Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8	8 years ago
George Joseph	66b8c454d1	Merge "app_playback.c: Use the timezonename parameter"	8 years ago
Joshua Colp	680c491a62	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	8 years ago
George Joseph	47084ad09e	Merge "app_queue: Add change priority of call"	8 years ago