30118 Commits (12aa25b2e150068abe3cac98d2d85498c3194ef0)

 

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Author	SHA1	Message	Date
Richard Mudgett	1ff580bb25	app_confbridge: ConfbridgeList event has standard channel shapshot headers. ... * Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977	8 years ago
Richard Mudgett	0cf7a9e0ca	app_confbridge: Add the Muted header to ConfbridgeJoin AMI event. ... ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34	8 years ago
Jenkins2	88e323a2a0	Merge "res_sorcery_realtime.c: Fix ref leak if object failed to apply." into 13	8 years ago
Jenkins2	2bb68c707f	Merge "manager.c: Fixed "(null):" header in AMI AsyncAGIEnd event" into 13	8 years ago
Jenkins2	d2df34f204	Merge "res_pjsip.c: Fix documentation typos." into 13	8 years ago
Jenkins2	65c8b7d6b4	Merge "manager_channels.c: Reordered ast_manager_build_channel_state_string_prefix()" into 13	8 years ago
Jenkins2	4bc5291878	Merge "res_pjsip_mwi.c: Fix null pointer crash" into 13	8 years ago
Jenkins2	34e3b51296	Merge "manager.c: Fix potential memory leak and corruption." into 13	8 years ago
Richard Mudgett	1adf90320d	endpoint identifiers: Some code cleanup. ... res_pjsip_endpoint_identifier_user.c: * Fix copy/paste error in find_endpoint(). We were using a constant "anonymous" string instead of the passed in endpoint_name when checking the transport domain for an endpoint match. * Eliminate RAII_VAR in find_endpoint(). * Remove always true check in find_transport_state_in_use(). * Remove useless CMD_STOP in find_transport_state_in_use(). res_pjsip_endpoint_identifier_anonymous.c: * Eliminate RAII_VAR in anonymous_identify(). * Remove always true check in find_transport_state_in_use(). * Remove useless CMD_STOP in find_transport_state_in_use(). Change-Id: I86924c31db5bd225ca0c1219c761b668c6f91189	8 years ago
Richard Mudgett	93a1ffc834	res_pjsip.c: Fix documentation typos. ... Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068	8 years ago
Richard Mudgett	5d16cefd63	res_sorcery_realtime.c: Fix ref leak if object failed to apply. ... Change-Id: I3c7106ff77009754725cee790eadf5da44154ab6	8 years ago
Jenkins2	8941934e0b	Merge "appdocsxml.xslt: Add Language to channel snapshot transformation" into 13	8 years ago
Sungtae Kim	dd9690f68c	manager.c: Fixed "(null):" header in AMI AsyncAGIEnd event ... * Changed to create ami_event string only when the given blob is not json_null(). * Fixed bad expression. ASTERISK-27621 Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f	8 years ago
Jenkins2	6b172f6f2b	Merge "bridge_softmix.c: Report not talking immediately when muted." into 13	8 years ago
Joshua Elson	8125b825d3	res_pjsip_mwi.c: Fix null pointer crash ... ASTERISK-27652 #close Change-Id: I78a0d38bfd8d0d82830f3d53da04872d6b67284d	8 years ago
Sean Bright	09edcd93aa	appdocsxml.xslt: Add Language to channel snapshot transformation ... Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8	8 years ago
Richard Mudgett	665444b772	manager.c: Fix potential memory leak and corruption. ... ast_str_append_event_header() could potentially leak and corrupt memory if the ast_str needed to expand to add the AMI event header. * Fixed to return error if the ast_str_append() failed. Change-Id: I92f36b855540743b208d76e274152ee2d758176d	8 years ago
Richard Mudgett	ef1aeb9437	manager_channels.c: Reordered ast_manager_build_channel_state_string_prefix() ... * Made not allocate memory if the channel snapshot is an internal channel. * Free memory earlier when no longer needed. Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38	8 years ago
Jenkins2	62d491527e	Merge "app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs." into 13	8 years ago
George Joseph	7debdd285c	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	8 years ago
Jenkins2	58ba7490a9	Merge "res_pjsip_registrar_expire: Mark module deprecated." into 13	8 years ago
Jenkins2	7cd9798a46	Merge "BuildSystem: Raise autoconf version requirement to 2.60a." into 13	8 years ago
Jenkins2	c9bff6be9b	Merge "res_pjsip_session: Prevent crash during shutdown." into 13	8 years ago
Jenkins2	994e62668b	Merge "app_voicemail: Avoid always true when using pointer address." into 13	8 years ago
Corey Farrell	698b28e9c7	res_pjsip_registrar_expire: Mark module deprecated. ... The functionality of this module was already moved to res_pjsip_registrar, mark it deprecated and add message to CHANGES. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813	8 years ago
Richard Mudgett	6520489952	bridge_softmix.c: Report not talking immediately when muted. ... Currently in app_confbridge if someone mutes a channel while that channel is talking, the talk detection code is suspended while the channel is muted. As far an an external observer is concerned, the muted channel's talk status is still "talking" even though the channel is not contributing audio to the conference bridge. When the channel is later unmuted, it takes the usual 'dsp_silence_threshold' option time to clear the talking status even though the channel may have stopped talking while the channel was muted. * In bridge_softmix.c, clear the talking status and report talking stopped if the channel was talking when the channel is muted. When the channel is unmuted and the channel is still talking then report the channel as talking since it is contributing audio to the bridge again. ASTERISK-27647 Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e	8 years ago
Richard Mudgett	4a337b1a76	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. ... The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	8 years ago
Richard Mudgett	2c4dde0ed0	res_pjsip_registrar.c: Fix compiler error. ... Need to include signal.h to define pthread_kill() and SIGURG. Change-Id: I10ae3aa4bf8e7386ac29ade78c0f2caed8e674fa	8 years ago
Jenkins2	d82da7afff	Merge "loader: Use ast_cli_completion_add for 'module load' completion." into 13	8 years ago
Jenkins2	ae236c5455	Merge "res_pjsip_registrar_expire: Refactor into res_pjsip_register" into 13	8 years ago
Jenkins2	cee39bf820	Merge "pbx_variables.c: Misc fixes in variable substitution." into 13	8 years ago
Jenkins2	0d105398f0	Merge "install_prereq: Update RHEL/CentOS/Fedora libraries." into 13	8 years ago
Corey Farrell	4c8c0e4c22	res_pjsip_session: Prevent crash during shutdown. ... pjproject does not have a function to reverse pjsip_inv_usage_init. This means we need to ignore any calls to the functions once shutdown is final. ASTERISK-27571 #close Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d	8 years ago
George Joseph	5b699c812e	res_pjsip_registrar_expire: Refactor into res_pjsip_register ... res_pjsip_registrar_expire remains as an empty module for now. Change-Id: Ib93698938bae548d2199cb542f3692d1a171239f	8 years ago
Jenkins2	897f7fb30e	Merge "Update sounds release to fix siren7 and siren14 files." into 13	8 years ago
Jenkins2	c956349667	Merge "core: Fix unused variable error in handle_show_sysinfo." into 13	8 years ago
Jenkins2	a1c7383d49	Merge "editline: Avoid shifting a negative signed value." into 13	8 years ago
Alexander Traud	967c70acb5	BuildSystem: Enable autotools in FreeBSD. ... In the current versions of FreeBSD, the apps of GNU autotools do not need to be called with a version anymore. The latest version can be invoked directly. Additionally, the script ./bootstrap.sh asked for autoconf 2.62 and automake 1.9, versions which are not available as port anymore. ASTERISK-27637 Change-Id: Id7b94b80e78cc943a40ba79b697e3f70019820a7	8 years ago
Jenkins2	5b3b627821	Merge "headers: Consistent use of typeof and/or __typeof__." into 13	8 years ago
Alexander Traud	dd6b8cd0b2	app_voicemail: Avoid always true when using pointer address. ... clang 4.0 warned about this. ASTERISK-27635 Change-Id: I213f230607d7fbe97c0f5f2d60da9cbf5a2d8231	8 years ago
Jenkins2	663f86a1a9	Merge "BuildSystem: Remove unused variables." into 13	8 years ago
Alexander Traud	8daed642c7	install_prereq: Update RHEL/CentOS/Fedora libraries. ... deleted automake git ncurses-devel pjproject-devel sqlite2-devel libsqlite3x-devel renamed radiusclient-ng-devel to radcli-devel gmime22-devel to gmime-dev added alsa-lib-devel bash binutils-devel bison doxygen flex hoard make pkgconfig speexdsp-devel uriparser-devel uw-imap-devel wget xmlstarlet zlib-devel ASTERISK-27599 Reported by: Said Masoud Change-Id: I05bb0af98ae532b2d5f37478e38b8f0762b1c035	8 years ago
Alexander Traud	f59fb7fec8	BuildSystem: Remove unused variables. ... Because of a copy-and-paste from the script build_tools/download_externals, the script build_tools/list_valid_installed_externals got its local variables. However in the latter, three variables were not used actually. Change-Id: I252de5a98c17ea54459174875357c22c2eebe8d5	8 years ago
Corey Farrell	154bccf147	loader: Use ast_cli_completion_add for 'module load' completion. ... This addresses all performance issues with 'module load' completion. In addition to using ast_cli_completion_add we stop using libedit's filename_completion_function, instead using ast_file_read_dir. This ensures all results are produced from a single call to opendir. Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134	8 years ago
Alexander Traud	4bb38022ea	core: Fix unused variable error in handle_show_sysinfo. ... The previous fix broke the case HAVE_SYSINFO = no HAVE_SYSCTL = yes HAVE_SWAPCTL = no which occurs on FreeBSD 11.1 for example. ASTERISK-26563 Change-Id: If77c39bc75f0b83a6c8a24ecb2fa69be8846160a	8 years ago
Alexander Traud	021168f06f	editline: Avoid shifting a negative signed value. ... clang 4.0 warned about this. ASTERISK-27630 Change-Id: Ie2725048c661c1792d8b1d498575144350b6e9ba	8 years ago
Alexander Traud	e998c906a7	headers: Consistent use of typeof and/or __typeof__. ... Because of a copy-and-paste error, the Asterisk project was using __typeof instead of typeof. It works because typeof, __typeof, and __typeof__ are supported by GCC, but here the escaped variant was not intended. Therefore, for consistence, we change this to typeof. Change-Id: I2a962c3e596e882f691a19345445b14571a5f07c	8 years ago
Richard Mudgett	87f8459410	Update sounds release to fix siren7 and siren14 files. ... ASTERISK-16172 Change-Id: I2fb564258cd4db0f35952ad48b8687355c2dcad3	8 years ago
Alexander Traud	6b771e346b	BuildSystem: Raise autoconf version requirement to 2.60a. ... AC_COMPUTE_INT requires at least autoconf 2.60a. This affects only those who contribute to Asterisk, only those who had to use the script ./bootstrap.sh. Furthermore, this change just makes sure nobody is using a too old autoconf. ASTERISK-16951 Change-Id: Ibca850e2fe0e77d935207bd959bacf7197d7f637	8 years ago
Alexander Traud	7ad6792011	install_prereq: Download latest Jansson. ... ASTERISK-27603 Change-Id: I65c587534c0ae364f063d68da1bed40bb3d5e8aa	8 years ago