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2017-12-21Merge "docs: Remove old API changes documentation" into 13Joshua Colp
2017-12-21Merge "app_festival: Fix fd leak on connection failure." into 13Jenkins2
2017-12-21Merge "bridge: Old channel video source not set to NULL after unref." into 13Jenkins2
2017-12-21Merge "core: Fix unused variable error in handle_show_sysinfo." into 13Joshua Colp
2017-12-20docs: Remove old API changes documentationSean Bright
Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a
2017-12-20manager.h: Bump AMI versionRichard Mudgett
Change-Id: I62e6ddeb261ef012687e1fb6734c554e2499b6bf
2017-12-20app_festival: Fix fd leak on connection failure.Corey Farrell
Change-Id: If5efaddcf735ff7d17e55c36cc1388946cee9e0f
2017-12-20bridge: Old channel video source not set to NULL after unref.Corey Farrell
The bridge holds onto the old channel video source after it's been released. This can lead to use after free errors. ASTERISK-27229 #close Change-Id: Ib2dab61677dd8a21f7ad53cdc9b8ca93297838b3
2017-12-20core: Fix unused variable error in handle_show_sysinfo.Corey Farrell
Apparently in OSX it's possible for OSX to HAVE_SYSCTL but not HAVE_SYSINFO or HAVE_SWAPCTL. In this case freeswap caused an unused variable error. ASTERISK-26563 #close Change-Id: I8ec5b1897b786cc1abaf62264aa75039eea05510
2017-12-20Merge "CLI: Address multiple issues." into 13Joshua Colp
2017-12-20Merge "bridge: Stop music on hold on adding an arbitrary channel to a ↵Jenkins2
bridge" into 13
2017-12-20Merge "res_rtp_asterisk: Avoid close the rtp/rtcp fd twice." into 13Jenkins2
2017-12-20Merge "core: Fix multiple trivial issues in the core." into 13Joshua Colp
2017-12-19Merge "chan_sip: Fix memory leaks." into 13Jenkins2
2017-12-19Merge "Remove constant conditionals (dead-code)." into 13Jenkins2
2017-12-19Merge "chan_console: Use correct parameter for 'set active'" into 13George Joseph
2017-12-19CLI: Address multiple issues.Corey Farrell
* listen uses the variable `s` for the result from ast_poll() then overwrites it with the result of accept(). Create a separate variable poll_result to avoid confusion since ast_poll does not return a file descriptor. * Resolve fd leak that would occur if setsockopt failed in listen. * Reserve an extra byte while processing completion results from remote daemon. This fixes a bug where completion processing used strstr() on a string that was not '\0' terminated. This was no risk to the Asterisk daemon, the bug was only reachable the remote console process. * Resolve leak in handle_showchan when the channel is not found. * Multiple leaks and a deadlock in pbx_config CLI completion. * Fix leaks in "manager show command". Change-Id: I8f633ceb1714867ae30ef4e421858f77c14485a9
2017-12-19res_rtp_asterisk: Avoid close the rtp/rtcp fd twice.Aaron An
When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before close the file descriptor. Close the FD twice will hangs the asterisk under heavy load. ASTERISK-27299 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: I870a072d73fd207463ac116ef97100addbc0820a
2017-12-19Merge "main/app: Fix leaks." into 13Jenkins2
2017-12-19bridge: Stop music on hold on adding an arbitrary channel to a bridgeIvan Poddubny
When a channel that is on hold gets added to a bridge by the Bridge AMI action or the dialplan application of the same name, music continues to play, causing "robotic sound". This commit adds a call to ast_moh_stop to stop the music. Also, it makes the AMI Park action use the right MOH class when the channel gets parked. Reported by: Zane Conkle ASTERISK-25079 #close Change-Id: I4b129c5a20c15e63968842460ac5a1a85903cf9f
2017-12-19chan_sip: Fix memory leaks.Corey Farrell
In change_redirecting_information variables we use ast_strlen_zero to see if a value should be saved. In the case where the value is not NULL but is a zero length string we leaked. handle_response_subscribe leaked a reference to the ccss monitor instance. Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f
2017-12-19Merge "func_callerid: Initialize app argument structures." into 13Jenkins2
2017-12-19Merge "rtp_engine.c: Eliminate rtcp_report_to_json() RAII_VAR usage." into 13Jenkins2
2017-12-19chan_console: Use correct parameter for 'set active'Oron Peled
chan_console supports multiple devices but the CLI only works on a single device. 'console set active' selects this device. Sadly that CLI picks the wrong command-line parameter and will only work for a device called 'active'. ASTERISK-27490 #close Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d
2017-12-19Merge "ast_json_pack(): Use safer json ref mechanism." into 13Jenkins2
2017-12-19Remove constant conditionals (dead-code).Corey Farrell
Some variables are set and never changed, making them constant. This means that code in the 'false' block of the conditional is unreachable. In chan_skinny and res_config_ldap I used preprocessor directive `#if 0` as I'm unsure if the unreachable code could be enabled in the future. Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
2017-12-19core: Fix multiple trivial issues in the core.Corey Farrell
* Fix small leaks in from error condition in translate.c. * Check new file descriptor is less than 0, not less than or equal. Change-Id: Id7782775486175c739e0c4bf3ea5e17e3f452a99
2017-12-18main/app: Fix leaks.Corey Farrell
* ast_linear_stream would leak a file descriptor if it failed to allocate lin. * ast_control_tone leaked zone and ts if ast_playtones_start failed. Additionally added whitespace to ast_linear_stream, pulled assignments out of conditionals for improved readability. Change-Id: I6d1a10cf9161b1529d939b9b2d63ea36d395b657
2017-12-18func_callerid: Initialize app argument structures.Corey Farrell
This module uses AST_DEFINE_APP_ARGS_TYPE to define struct's instead of directly using AST_DECLARE_APP_ARGS. Initialize the variables declared in this way. Change-Id: If97fbdd8d63a204e2efd498a192effc14e90fb31
2017-12-18app_voicemail: Fix memory management issues.Corey Farrell
* mwi_sub_event_cb: mwist leaked on separate_mailbox failure. * add_email_attachment: A reference to sox_gain_tmpdir was used after the storage was out of scope. Change-Id: I6282c542ff7b82fa091177a912d11234a8b00a30
2017-12-18ast_json_pack(): Use safer json ref mechanism.Richard Mudgett
Change-Id: I49204db2e57ae96eee43909c18ed007c09ac817e
2017-12-18rtp_engine.c: Eliminate rtcp_report_to_json() RAII_VAR usage.Richard Mudgett
Change-Id: I58a22c2ca82e91d7537409b7b3af2d735827a54d
2017-12-18Merge "CLI: Fix 'core set debug channel' completion bug." into 13Joshua Colp
2017-12-18Merge "CLI: Remove special handling of 'core set verbose' from rasterisk." ↵Jenkins2
into 13
2017-12-18Merge "aco: Minimize use of regex." into 13Jenkins2
2017-12-18Merge "aco: Create ways to minimize use of regex." into 13Jenkins2
2017-12-18CLI: Fix 'core set debug channel' completion bug.Corey Farrell
The completion generator is missing a return so typing "core set debug all off <tab>" causes the command to actually execute. Change-Id: Ibf6462088a74eee66967732b50445783ebefc20b
2017-12-18Merge "confbridge: Clarify mute sound documentation." into 13Joshua Colp
2017-12-18Merge "loader: Use vector to build apha sorted module lists." into 13George Joseph
2017-12-18Merge "loader: Replace priority heap with vector." into 13George Joseph
2017-12-18Merge "chan_pjsip.c: Improve ast_request() diagnostic msgs." into 13Jenkins2
2017-12-18Merge "chan_sip: Add security event for calls to invalid extension." into 13Jenkins2
2017-12-18Merge "cdr: Minor optimizations." into 13Jenkins2
2017-12-18Merge "loader: Rework of load_dynamic_module." into 13George Joseph
2017-12-18confbridge: Clarify mute sound documentation.Joshua Colp
The mute/unmute sounds are only played when the action is initiated using the DTMF menu. ASTERISK-24756 Change-Id: I55b3dd5bc166096bf5e2f547ddd0ce355f36e3dc
2017-12-18Merge "loader: Minor fix to module registration." into 13Jenkins2
2017-12-18app_transfer: Remove LOCAL from documentation.Joshua Colp
The Local channel has never supported app_transfer from what I can see so remove it from the documentation. ASTERISK-25649 Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9
2017-12-15chan_pjsip.c: Improve ast_request() diagnostic msgs.Richard Mudgett
Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and disable_multi_domain=no results in a misleading empty endpoint name message. The message should say the endpoint was not found. * Added missing endpoint not found message. * Added more information to the empty endpoint name msgs if available. * Eliminated RAII_VAR in request(). Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4
2017-12-15Merge "res_smdi: Fix shutdown ref." into 13Jenkins2
2017-12-15Merge "res_rtp_asterisk.c: Disable packet flood detection for video ↵Jenkins2
streams." into 13